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<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
<?xml-stylesheet type="text/xsl" href="appdocsxml.xslt"?>
<docs xmlns:xi="http://www.w3.org/2001/XInclude">
<function name="PJSIP_DIAL_CONTACTS" language="en_US">
<synopsis>
Return a dial string for dialing all contacts on an AOR.
</synopsis>
<syntax>
<parameter name="endpoint" required="true">
<para>Name of the endpoint</para>
</parameter>
<parameter name="aor" required="false">
<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
</parameter>
<parameter name="request_user" required="false">
<para>Optional request user to use in the request URI</para>
</parameter>
</syntax>
<description>
<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
</description>
</function>
<function name="PJSIP_MEDIA_OFFER" language="en_US">
<synopsis>
Media and codec offerings to be set on an outbound SIP channel prior to dialing.
</synopsis>
<syntax>
<parameter name="media" required="true">
<para>types of media offered</para>
</parameter>
</syntax>
<description>
<para>When read, returns the codecs offered based upon the media choice.</para>
<para>When written, sets the codecs to offer when an outbound dial attempt is made,
or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
</para>
</description>
<see-also>
<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
</see-also>
</function>
<function name="PJSIP_DTMF_MODE" language="en_US">
<synopsis>
Get or change the DTMF mode for a SIP call.
</synopsis>
<syntax>
</syntax>
<description>
<para>When read, returns the current DTMF mode</para>
<para>When written, sets the current DTMF mode</para>
<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
</description>
</function>
<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
<synopsis>
W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
</synopsis>
<syntax>
<parameter name="update_type" required="false">
<para>The type of update to send. Default is <literal>invite</literal>.</para>
<enumlist>
<enum name="invite">
<para>Send the session refresh as a re-INVITE.</para>
</enum>
<enum name="update">
<para>Send the session refresh as an UPDATE.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function will cause the PJSIP stack to immediately refresh
the media session for the channel. This will be done using either a
re-INVITE (default) or an UPDATE request.
</para>
<para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
dialplan function, as it allows the formats in use on a channel to be
re-negotiated after call setup.</para>
<warning>
<para>The formats the endpoint supports are <emphasis>not</emphasis>
checked or enforced by this function. Using this function to offer
formats not supported by the endpoint <emphasis>may</emphasis> result
in a loss of media.</para>
</warning>
<example title="Re-negotiate format to g722">
; Within some existing extension on an answered channel
same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
</example>
</description>
<see-also>
<ref type="function">PJSIP_MEDIA_OFFER</ref>
</see-also>
</function>
<info name="CHANNEL" language="en_US" tech="PJSIP">
<enumlist>
<enum name="rtp">
<para>R/O Retrieve media related information.</para>
<parameter name="type" required="true">
<para>When <replaceable>rtp</replaceable> is specified, the
<literal>type</literal> parameter must be provided. It specifies
which RTP parameter to read.</para>
<enumlist>
<enum name="src">
<para>Retrieve the local address for RTP.</para>
</enum>
<enum name="dest">
<para>Retrieve the remote address for RTP.</para>
</enum>
<enum name="direct">
<para>If direct media is enabled, this address is the remote address
used for RTP.</para>
</enum>
<enum name="secure">
<para>Whether or not the media stream is encrypted.</para>
<enumlist>
<enum name="0">
<para>The media stream is not encrypted.</para>
</enum>
<enum name="1">
<para>The media stream is encrypted.</para>
</enum>
</enumlist>
</enum>
<enum name="hold">
<para>Whether or not the media stream is currently restricted
due to a call hold.</para>
<enumlist>
<enum name="0">
<para>The media stream is not held.</para>
</enum>
<enum name="1">
<para>The media stream is held.</para>
</enum>
</enumlist>
</enum>
</enumlist>
</parameter>
<parameter name="media_type" required="false">
<para>When <replaceable>rtp</replaceable> is specified, the
<literal>media_type</literal> parameter may be provided. It specifies
which media stream the chosen RTP parameter should be retrieved
from.</para>
<enumlist>
<enum name="audio">
<para>Retrieve information from the audio media stream.</para>
<note>
<para>If not specified, <literal>audio</literal> is used
by default.</para>
</note>
</enum>
<enum name="video">
<para>Retrieve information from the video media stream.</para>
</enum>
</enumlist>
</parameter>
</enum>
<enum name="rtcp">
<para>R/O Retrieve RTCP statistics.</para>
<parameter name="statistic" required="true">
<para>When <replaceable>rtcp</replaceable> is specified, the
<literal>statistic</literal> parameter must be provided. It specifies
which RTCP statistic parameter to read.</para>
<enumlist>
<enum name="all">
<para>Retrieve a summary of all RTCP statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="ssrc">
<para>Our Synchronization Source identifier</para>
</enum>
<enum name="themssrc">
<para>Their Synchronization Source identifier</para>
</enum>
<enum name="lp">
<para>Our lost packet count</para>
</enum>
<enum name="rxjitter">
<para>Received packet jitter</para>
</enum>
<enum name="rxcount">
<para>Received packet count</para>
</enum>
<enum name="txjitter">
<para>Transmitted packet jitter</para>
</enum>
<enum name="txcount">
<para>Transmitted packet count</para>
</enum>
<enum name="rlp">
<para>Remote lost packet count</para>
</enum>
<enum name="rtt">
<para>Round trip time</para>
</enum>
</enumlist>
</enum>
<enum name="all_jitter">
<para>Retrieve a summary of all RTCP Jitter statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrxjitter">
<para>Our minimum jitter</para>
</enum>
<enum name="maxrxjitter">
<para>Our max jitter</para>
</enum>
<enum name="avgrxjitter">
<para>Our average jitter</para>
</enum>
<enum name="stdevrxjitter">
<para>Our jitter standard deviation</para>
</enum>
<enum name="reported_minjitter">
<para>Their minimum jitter</para>
</enum>
<enum name="reported_maxjitter">
<para>Their max jitter</para>
</enum>
<enum name="reported_avgjitter">
<para>Their average jitter</para>
</enum>
<enum name="reported_stdevjitter">
<para>Their jitter standard deviation</para>
</enum>
</enumlist>
</enum>
<enum name="all_loss">
<para>Retrieve a summary of all RTCP packet loss statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrxlost">
<para>Our minimum lost packets</para>
</enum>
<enum name="maxrxlost">
<para>Our max lost packets</para>
</enum>
<enum name="avgrxlost">
<para>Our average lost packets</para>
</enum>
<enum name="stdevrxlost">
<para>Our lost packets standard deviation</para>
</enum>
<enum name="reported_minlost">
<para>Their minimum lost packets</para>
</enum>
<enum name="reported_maxlost">
<para>Their max lost packets</para>
</enum>
<enum name="reported_avglost">
<para>Their average lost packets</para>
</enum>
<enum name="reported_stdevlost">
<para>Their lost packets standard deviation</para>
</enum>
</enumlist>
</enum>
<enum name="all_rtt">
<para>Retrieve a summary of all RTCP round trip time information.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrtt">
<para>Minimum round trip time</para>
</enum>
<enum name="maxrtt">
<para>Maximum round trip time</para>
</enum>
<enum name="avgrtt">
<para>Average round trip time</para>
</enum>
<enum name="stdevrtt">
<para>Standard deviation round trip time</para>
</enum>
</enumlist>
</enum>
<enum name="txcount">
<para>Transmitted packet count</para>
</enum>
<enum name="rxcount">
<para>Received packet count</para>
</enum>
<enum name="txjitter">
<para>Transmitted packet jitter</para>
</enum>
<enum name="rxjitter">
<para>Received packet jitter</para>
</enum>
<enum name="remote_maxjitter">
<para>Their max jitter</para>
</enum>
<enum name="remote_minjitter">
<para>Their minimum jitter</para>
</enum>
<enum name="remote_normdevjitter">
<para>Their average jitter</para>
</enum>
<enum name="remote_stdevjitter">
<para>Their jitter standard deviation</para>
</enum>
<enum name="local_maxjitter">
<para>Our max jitter</para>
</enum>
<enum name="local_minjitter">
<para>Our minimum jitter</para>
</enum>
<enum name="local_normdevjitter">
<para>Our average jitter</para>
</enum>
<enum name="local_stdevjitter">
<para>Our jitter standard deviation</para>
</enum>
<enum name="txploss">
<para>Transmitted packet loss</para>
</enum>
<enum name="rxploss">
<para>Received packet loss</para>
</enum>
<enum name="remote_maxrxploss">
<para>Their max lost packets</para>
</enum>
<enum name="remote_minrxploss">
<para>Their minimum lost packets</para>
</enum>
<enum name="remote_normdevrxploss">
<para>Their average lost packets</para>
</enum>
<enum name="remote_stdevrxploss">
<para>Their lost packets standard deviation</para>
</enum>
<enum name="local_maxrxploss">
<para>Our max lost packets</para>
</enum>
<enum name="local_minrxploss">
<para>Our minimum lost packets</para>
</enum>
<enum name="local_normdevrxploss">
<para>Our average lost packets</para>
</enum>
<enum name="local_stdevrxploss">
<para>Our lost packets standard deviation</para>
</enum>
<enum name="rtt">
<para>Round trip time</para>
</enum>
<enum name="maxrtt">
<para>Maximum round trip time</para>
</enum>
<enum name="minrtt">
<para>Minimum round trip time</para>
</enum>
<enum name="normdevrtt">
<para>Average round trip time</para>
</enum>
<enum name="stdevrtt">
<para>Standard deviation round trip time</para>
</enum>
<enum name="local_ssrc">
<para>Our Synchronization Source identifier</para>
</enum>
<enum name="remote_ssrc">
<para>Their Synchronization Source identifier</para>
</enum>
</enumlist>
</parameter>
<parameter name="media_type" required="false">
<para>When <replaceable>rtcp</replaceable> is specified, the
<literal>media_type</literal> parameter may be provided. It specifies
which media stream the chosen RTCP parameter should be retrieved
from.</para>
<enumlist>
<enum name="audio">
<para>Retrieve information from the audio media stream.</para>
<note>
<para>If not specified, <literal>audio</literal> is used
by default.</para>
</note>
</enum>
<enum name="video">
<para>Retrieve information from the video media stream.</para>
</enum>
</enumlist>
</parameter>
</enum>
<enum name="endpoint">
<para>R/O The name of the endpoint associated with this channel.
Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
further endpoint related information.</para>
</enum>
<enum name="contact">
<para>R/O The name of the contact associated with this channel.
Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
further contact related information. Note this may not be present and if so
is only available on outgoing legs.</para>
</enum>
<enum name="aor">
<para>R/O The name of the AOR associated with this channel.
Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
further AOR related information. Note this may not be present and if so
is only available on outgoing legs.</para>
</enum>
<enum name="pjsip">
<para>R/O Obtain information about the current PJSIP channel and its
session.</para>
<parameter name="type" required="true">
<para>When <replaceable>pjsip</replaceable> is specified, the
<literal>type</literal> parameter must be provided. It specifies
which signalling parameter to read.</para>
<enumlist>
<enum name="call-id">
<para>The SIP call-id.</para>
</enum>
<enum name="secure">
<para>Whether or not the signalling uses a secure transport.</para>
<enumlist>
<enum name="0">
<para>The signalling uses a non-secure transport.</para>
</enum>
<enum name="1">
<para>The signalling uses a secure transport.</para>
</enum>
</enumlist>
</enum>
<enum name="target_uri">
<para>The contact URI where requests are sent.</para>
</enum>
<enum name="local_uri">
<para>The local URI.</para>
</enum>
<enum name="local_tag">
<para>Tag in From header</para>
</enum>
<enum name="remote_uri">
<para>The remote URI.</para>
</enum>
<enum name="remote_tag">
<para>Tag in To header</para>
</enum>
<enum name="request_uri">
<para>The request URI of the incoming <literal>INVITE</literal>
associated with the creation of this channel.</para>
</enum>
<enum name="t38state">
<para>The current state of any T.38 fax on this channel.</para>
<enumlist>
<enum name="DISABLED">
<para>T.38 faxing is disabled on this channel.</para>
</enum>
<enum name="LOCAL_REINVITE">
<para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para>
</enum>
<enum name="REMOTE_REINVITE">
<para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para>
</enum>
<enum name="ENABLED">
<para>A T.38 fax session has been enabled.</para>
</enum>
<enum name="REJECTED">
<para>A T.38 fax session was attempted but was rejected.</para>
</enum>
</enumlist>
</enum>
<enum name="local_addr">
<para>On inbound calls, the full IP address and port number that
the <literal>INVITE</literal> request was received on. On outbound
calls, the full IP address and port number that the <literal>INVITE</literal>
request was transmitted from.</para>
</enum>
<enum name="remote_addr">
<para>On inbound calls, the full IP address and port number that
the <literal>INVITE</literal> request was received from. On outbound
calls, the full IP address and port number that the <literal>INVITE</literal>
request was transmitted to.</para>
</enum>
</enumlist>
</parameter>
</enum>
</enumlist>
</info>
<info name="CHANNEL_EXAMPLES" language="en_US" tech="PJSIP">
<example title="PJSIP specific CHANNEL examples">
; Log the current Call-ID
same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})
; Log the destination address of the audio stream
same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})
; Store the round-trip time associated with a
; video stream in the CDR field video-rtt
same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})
</example>
</info>
<info name="CHANNEL" language="en_US" tech="SIP">
<enumlist>
<enum name="peerip">
<para>R/O Get the IP address of the peer.</para>
</enum>
<enum name="recvip">
<para>R/O Get the source IP address of the peer.</para>
</enum>
<enum name="recvport">
<para>R/O Get the source port of the peer.</para>
</enum>
<enum name="from">
<para>R/O Get the URI from the From: header.</para>
</enum>
<enum name="uri">
<para>R/O Get the URI from the Contact: header.</para>
</enum>
<enum name="ruri">
<para>R/O Get the Request-URI from the INVITE header.</para>
</enum>
<enum name="useragent">
<para>R/O Get the useragent.</para>
</enum>
<enum name="peername">
<para>R/O Get the name of the peer.</para>
</enum>
<enum name="t38passthrough">
<para>R/O <literal>1</literal> if T38 is offered or enabled in this channel,
otherwise <literal>0</literal>
</para>
</enum>
<enum name="rtpqos">
<para>R/O Get QOS information about the RTP stream</para>
<para> This option takes two additional arguments:</para>
<para> Argument 1:</para>
<para>
<literal>audio</literal> Get data about the audio stream</para>
<para>
<literal>video</literal> Get data about the video stream</para>
<para>
<literal>text</literal> Get data about the text stream</para>
<para> Argument 2:</para>
<para>
<literal>local_ssrc</literal> Local SSRC (stream ID)</para>
<para>
<literal>local_lostpackets</literal> Local lost packets</para>
<para>
<literal>local_jitter</literal> Local calculated jitter</para>
<para>
<literal>local_maxjitter</literal> Local calculated jitter (maximum)</para>
<para>
<literal>local_minjitter</literal> Local calculated jitter (minimum)</para>
<para>
<literal>local_normdevjitter</literal>Local calculated jitter (normal deviation)</para>
<para>
<literal>local_stdevjitter</literal> Local calculated jitter (standard deviation)</para>
<para>
<literal>local_count</literal> Number of received packets</para>
<para>
<literal>remote_ssrc</literal> Remote SSRC (stream ID)</para>
<para>
<literal>remote_lostpackets</literal>Remote lost packets</para>
<para>
<literal>remote_jitter</literal> Remote reported jitter</para>
<para>
<literal>remote_maxjitter</literal> Remote calculated jitter (maximum)</para>
<para>
<literal>remote_minjitter</literal> Remote calculated jitter (minimum)</para>
<para>
<literal>remote_normdevjitter</literal>Remote calculated jitter (normal deviation)</para>
<para>
<literal>remote_stdevjitter</literal>Remote calculated jitter (standard deviation)</para>
<para>
<literal>remote_count</literal> Number of transmitted packets</para>
<para>
<literal>rtt</literal> Round trip time</para>
<para>
<literal>maxrtt</literal> Round trip time (maximum)</para>
<para>
<literal>minrtt</literal> Round trip time (minimum)</para>
<para>
<literal>normdevrtt</literal> Round trip time (normal deviation)</para>
<para>
<literal>stdevrtt</literal> Round trip time (standard deviation)</para>
<para>
<literal>all</literal> All statistics (in a form suited to logging,
but not for parsing)</para>
</enum>
<enum name="rtpdest">
<para>R/O Get remote RTP destination information.</para>
<para> This option takes one additional argument:</para>
<para> Argument 1:</para>
<para>
<literal>audio</literal> Get audio destination</para>
<para>
<literal>video</literal> Get video destination</para>
<para>
<literal>text</literal> Get text destination</para>
<para> Defaults to <literal>audio</literal> if unspecified.</para>
</enum>
<enum name="rtpsource">
<para>R/O Get source RTP destination information.</para>
<para> This option takes one additional argument:</para>
<para> Argument 1:</para>
<para>
<literal>audio</literal> Get audio destination</para>
<para>
<literal>video</literal> Get video destination</para>
<para>
<literal>text</literal> Get text destination</para>
<para> Defaults to <literal>audio</literal> if unspecified.</para>
</enum>
</enumlist>
</info>
<application name="DAHDISendKeypadFacility" language="en_US">
<synopsis>
Send digits out of band over a PRI.
</synopsis>
<syntax>
<parameter name="digits" required="true" />
</syntax>
<description>
<para>This application will send the given string of digits in a Keypad
Facility IE over the current channel.</para>
</description>
</application>
<application name="DAHDISendCallreroutingFacility" language="en_US">
<synopsis>
Send an ISDN call rerouting/deflection facility message.
</synopsis>
<syntax argsep=",">
<parameter name="destination" required="true">
<para>Destination number.</para>
</parameter>
<parameter name="original">
<para>Original called number.</para>
</parameter>
<parameter name="reason">
<para>Diversion reason, if not specified defaults to <literal>unknown</literal></para>
</parameter>
</syntax>
<description>
<para>This application will send an ISDN switch specific call
rerouting/deflection facility message over the current channel.
Supported switches depend upon the version of libpri in use.</para>
</description>
</application>
<application name="DAHDIAcceptR2Call" language="en_US">
<synopsis>
Accept an R2 call if its not already accepted (you still need to answer it)
</synopsis>
<syntax>
<parameter name="charge" required="true">
<para>Yes or No.</para>
<para>Whether you want to accept the call with charge or without charge.</para>
</parameter>
</syntax>
<description>
<para>This application will Accept the R2 call either with charge or no charge.</para>
</description>
</application>
<info name="CHANNEL" language="en_US" tech="DAHDI">
<enumlist>
<enum name="dahdi_channel">
<para>R/O DAHDI channel related to this channel.</para>
</enum>
<enum name="dahdi_span">
<para>R/O DAHDI span related to this channel.</para>
</enum>
<enum name="dahdi_type">
<para>R/O DAHDI channel type, one of:</para>
<enumlist>
<enum name="analog" />
<enum name="mfc/r2" />
<enum name="pri" />
<enum name="pseudo" />
<enum name="ss7" />
</enumlist>
</enum>
<enum name="keypad_digits">
<para>R/O PRI Keypad digits that came in with the SETUP message.</para>
</enum>
<enum name="reversecharge">
<para>R/O PRI Reverse Charging Indication, one of:</para>
<enumlist>
<enum name="-1">
<para>None</para>
</enum>
<enum name=" 1">
<para>Reverse Charging Requested</para>
</enum>
</enumlist>
</enum>
<enum name="no_media_path">
<para>R/O PRI Nonzero if the channel has no B channel.
The channel is either on hold or a call waiting call.</para>
</enum>
<enum name="buffers">
<para>W/O Change the channel's buffer policy (for the current call only)</para>
<para>This option takes two arguments:</para>
<para> Number of buffers,</para>
<para> Buffer policy being one of:</para>
<para>
<literal>full</literal>
</para>
<para>
<literal>immediate</literal>
</para>
<para>
<literal>half</literal>
</para>
</enum>
<enum name="echocan_mode">
<para>W/O Change the configuration of the active echo
canceller on the channel (if any), for the current call
only.</para>
<para>Possible values are:</para>
<para>
<literal>on</literal> Normal mode (the echo canceller is actually reinitalized)</para>
<para>
<literal>off</literal> Disabled</para>
<para>
<literal>fax</literal> FAX/data mode (NLP disabled if possible, otherwise
completely disabled)</para>
<para>
<literal>voice</literal> Voice mode (returns from FAX mode, reverting the changes that were made)</para>
</enum>
</enumlist>
</info>
<manager name="DAHDITransfer" language="en_US">
<synopsis>
Transfer DAHDI Channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel" required="true">
<para>DAHDI channel number to transfer.</para>
</parameter>
</syntax>
<description>
<para>Simulate a flash hook event by the user connected to the channel.</para>
<note>
<para>Valid only for analog channels.</para>
</note>
</description>
</manager>
<manager name="DAHDIHangup" language="en_US">
<synopsis>
Hangup DAHDI Channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel" required="true">
<para>DAHDI channel number to hangup.</para>
</parameter>
</syntax>
<description>
<para>Simulate an on-hook event by the user connected to the channel.</para>
<note>
<para>Valid only for analog channels.</para>
</note>
</description>
</manager>
<manager name="DAHDIDialOffhook" language="en_US">
<synopsis>
Dial over DAHDI channel while offhook.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel" required="true">
<para>DAHDI channel number to dial digits.</para>
</parameter>
<parameter name="Number" required="true">
<para>Digits to dial.</para>
</parameter>
</syntax>
<description>
<para>Generate DTMF control frames to the bridged peer.</para>
</description>
</manager>
<manager name="DAHDIDNDon" language="en_US">
<synopsis>
Toggle DAHDI channel Do Not Disturb status ON.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel" required="true">
<para>DAHDI channel number to set DND on.</para>
</parameter>
</syntax>
<description>
<para>Equivalent to the CLI command "dahdi set dnd <variable>channel</variable> on".</para>
<note>
<para>Feature only supported by analog channels.</para>
</note>
</description>
</manager>
<manager name="DAHDIDNDoff" language="en_US">
<synopsis>
Toggle DAHDI channel Do Not Disturb status OFF.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel" required="true">
<para>DAHDI channel number to set DND off.</para>
</parameter>
</syntax>
<description>
<para>Equivalent to the CLI command "dahdi set dnd <variable>channel</variable> off".</para>
<note>
<para>Feature only supported by analog channels.</para>
</note>
</description>
</manager>
<manager name="DAHDIShowChannels" language="en_US">
<synopsis>
Show status of DAHDI channels.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="DAHDIChannel">
<para>Specify the specific channel number to show. Show all channels if zero or not present.</para>
</parameter>
</syntax>
<description>
<para>Similar to the CLI command "dahdi show channels".</para>
</description>
</manager>
<manager name="DAHDIRestart" language="en_US">
<synopsis>
Fully Restart DAHDI channels (terminates calls).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Equivalent to the CLI command "dahdi restart".</para>
</description>
</manager>
<manager name="PRIShowSpans" language="en_US">
<synopsis>
Show status of PRI spans.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Span">
<para>Specify the specific span to show. Show all spans if zero or not present.</para>
</parameter>
</syntax>
<description>
<para>Similar to the CLI command "pri show spans".</para>
</description>
</manager>
<manager name="PRIDebugSet" language="en_US">
<synopsis>
Set PRI debug levels for a span
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Span" required="true">
<para>Which span to affect.</para>
</parameter>
<parameter name="Level" required="true">
<para>What debug level to set. May be a numerical value or a text value from the list below</para>
<enumlist>
<enum name="off" />
<enum name="on" />
<enum name="hex" />
<enum name="intense" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Equivalent to the CLI command "pri set debug &lt;level&gt; span &lt;span&gt;".</para>
</description>
</manager>
<manager name="PRIDebugFileSet" language="en_US">
<synopsis>
Set the file used for PRI debug message output
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="File" required="true">
<para>Path of file to write debug output.</para>
</parameter>
</syntax>
<description>
<para>Equivalent to the CLI command "pri set debug file &lt;output-file&gt;"</para>
</description>
</manager>
<manager name="PRIDebugFileUnset" language="en_US">
<synopsis>
Disables file output for PRI debug messages
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
</manager>
<managerEvent language="en_US" name="AlarmClear">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when an alarm is cleared on a DAHDI channel.</synopsis>
<syntax>
<parameter name="DAHDIChannel">
<para>The DAHDI channel on which the alarm was cleared.</para>
<note>
<para>This is not an Asterisk channel identifier.</para>
</note>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SpanAlarmClear">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when an alarm is cleared on a DAHDI span.</synopsis>
<syntax>
<parameter name="Span">
<para>The span on which the alarm was cleared.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="DNDState">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when the Do Not Disturb state is changed on a DAHDI channel.</synopsis>
<syntax>
<parameter name="DAHDIChannel">
<para>The DAHDI channel on which DND status changed.</para>
<note>
<para>This is not an Asterisk channel identifier.</para>
</note>
</parameter>
<parameter name="Status">
<enumlist>
<enum name="enabled"/>
<enum name="disabled"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Alarm">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when an alarm is set on a DAHDI channel.</synopsis>
<syntax>
<parameter name="DAHDIChannel">
<para>The channel on which the alarm occurred.</para>
<note>
<para>This is not an Asterisk channel identifier.</para>
</note>
</parameter>
<parameter name="Alarm">
<para>A textual description of the alarm that occurred.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SpanAlarm">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when an alarm is set on a DAHDI span.</synopsis>
<syntax>
<parameter name="Span">
<para>The span on which the alarm occurred.</para>
</parameter>
<parameter name="Alarm">
<para>A textual description of the alarm that occurred.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="DAHDIChannel">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a DAHDI channel is created or an underlying technology is associated with a DAHDI channel.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="DAHDISpan">
<para>The DAHDI span associated with this channel.</para>
</parameter>
<parameter name="DAHDIChannel">
<para>The DAHDI channel associated with this channel.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<application name="SIPDtmfMode" language="en_US">
<synopsis>
Change the dtmfmode for a SIP call.
</synopsis>
<syntax>
<parameter name="mode" required="true">
<enumlist>
<enum name="inband" />
<enum name="info" />
<enum name="rfc2833" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Changes the dtmfmode for a SIP call.</para>
</description>
</application>
<application name="SIPAddHeader" language="en_US">
<synopsis>
Add a SIP header to the outbound call.
</synopsis>
<syntax argsep=":">
<parameter name="Header" required="true" />
<parameter name="Content" required="true" />
</syntax>
<description>
<para>Adds a header to a SIP call placed with DIAL.</para>
<para>Remember to use the X-header if you are adding non-standard SIP
headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
Adding the wrong headers may jeopardize the SIP dialog.</para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<application name="SIPRemoveHeader" language="en_US">
<synopsis>
Remove SIP headers previously added with SIPAddHeader
</synopsis>
<syntax>
<parameter name="Header" required="false" />
</syntax>
<description>
<para>SIPRemoveHeader() allows you to remove headers which were previously
added with SIPAddHeader(). If no parameter is supplied, all previously added
headers will be removed. If a parameter is supplied, only the matching headers
will be removed.</para>
<para>For example you have added these 2 headers:</para>
<para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
<para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
<para></para>
<para>// remove all headers</para>
<para>SIPRemoveHeader();</para>
<para>// remove all P- headers</para>
<para>SIPRemoveHeader(P-);</para>
<para>// remove only the PAI header (note the : at the end)</para>
<para>SIPRemoveHeader(P-Asserted-Identity:);</para>
<para></para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<application name="SIPSendCustomINFO" language="en_US">
<synopsis>
Send a custom INFO frame on specified channels.
</synopsis>
<syntax>
<parameter name="Data" required="true" />
<parameter name="UserAgent" required="false" />
</syntax>
<description>
<para>SIPSendCustomINFO() allows you to send a custom INFO message on all
active SIP channels or on channels with the specified User Agent. This
application is only available if TEST_FRAMEWORK is defined.</para>
</description>
</application>
<function name="SIP_HEADER" language="en_US">
<synopsis>
Gets the specified SIP header from an incoming INVITE message.
</synopsis>
<syntax>
<parameter name="name" required="true" />
<parameter name="number">
<para>If not specified, defaults to <literal>1</literal>.</para>
</parameter>
</syntax>
<description>
<para>Since there are several headers (such as Via) which can occur multiple
times, SIP_HEADER takes an optional second argument to specify which header with
that name to retrieve. Headers start at offset <literal>1</literal>.</para>
<para>Please observe that contents of the SDP (an attachment to the
SIP request) can't be accessed with this function.</para>
</description>
</function>
<function name="SIPPEER" language="en_US">
<synopsis>
Gets SIP peer information.
</synopsis>
<syntax>
<parameter name="peername" required="true" />
<parameter name="item">
<enumlist>
<enum name="ip">
<para>(default) The IP address.</para>
</enum>
<enum name="port">
<para>The port number.</para>
</enum>
<enum name="mailbox">
<para>The configured mailbox.</para>
</enum>
<enum name="context">
<para>The configured context.</para>
</enum>
<enum name="expire">
<para>The epoch time of the next expire.</para>
</enum>
<enum name="dynamic">
<para>Is it dynamic? (yes/no).</para>
</enum>
<enum name="callerid_name">
<para>The configured Caller ID name.</para>
</enum>
<enum name="callerid_num">
<para>The configured Caller ID number.</para>
</enum>
<enum name="callgroup">
<para>The configured Callgroup.</para>
</enum>
<enum name="pickupgroup">
<para>The configured Pickupgroup.</para>
</enum>
<enum name="namedcallgroup">
<para>The configured Named Callgroup.</para>
</enum>
<enum name="namedpickupgroup">
<para>The configured Named Pickupgroup.</para>
</enum>
<enum name="codecs">
<para>The configured codecs.</para>
</enum>
<enum name="status">
<para>Status (if qualify=yes).</para>
</enum>
<enum name="regexten">
<para>Extension activated at registration.</para>
</enum>
<enum name="limit">
<para>Call limit (call-limit).</para>
</enum>
<enum name="busylevel">
<para>Configured call level for signalling busy.</para>
</enum>
<enum name="curcalls">
<para>Current amount of calls. Only available if call-limit is set.</para>
</enum>
<enum name="language">
<para>Default language for peer.</para>
</enum>
<enum name="accountcode">
<para>Account code for this peer.</para>
</enum>
<enum name="useragent">
<para>Current user agent header used by peer.</para>
</enum>
<enum name="maxforwards">
<para>The value used for SIP loop prevention in outbound requests</para>
</enum>
<enum name="chanvar[name]">
<para>A channel variable configured with setvar for this peer.</para>
</enum>
<enum name="codec[x]">
<para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description></description>
</function>
<function name="CHECKSIPDOMAIN" language="en_US">
<synopsis>
Checks if domain is a local domain.
</synopsis>
<syntax>
<parameter name="domain" required="true" />
</syntax>
<description>
<para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
as a local SIP domain that this Asterisk server is configured to handle.
Returns the domain name if it is locally handled, otherwise an empty string.
Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
</description>
</function>
<manager name="SIPpeers" language="en_US">
<synopsis>
List SIP peers (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists SIP peers in text format with details on current status.
<literal>Peerlist</literal> will follow as separate events, followed by a final event called
<literal>PeerlistComplete</literal>.</para>
</description>
</manager>
<manager name="SIPshowpeer" language="en_US">
<synopsis>
show SIP peer (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Peer" required="true">
<para>The peer name you want to check.</para>
</parameter>
</syntax>
<description>
<para>Show one SIP peer with details on current status.</para>
</description>
</manager>
<manager name="SIPqualifypeer" language="en_US">
<synopsis>
Qualify SIP peers.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Peer" required="true">
<para>The peer name you want to qualify.</para>
</parameter>
</syntax>
<description>
<para>Qualify a SIP peer.</para>
</description>
<see-also>
<ref type="managerEvent">SIPQualifyPeerDone</ref>
</see-also>
</manager>
<manager name="SIPshowregistry" language="en_US">
<synopsis>
Show SIP registrations (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists all registration requests and status. Registrations will follow as separate
events followed by a final event called <literal>RegistrationsComplete</literal>.</para>
</description>
</manager>
<manager name="SIPnotify" language="en_US">
<synopsis>
Send a SIP notify.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Peer to receive the notify.</para>
</parameter>
<parameter name="Variable" required="true">
<para>At least one variable pair must be specified.
<replaceable>name</replaceable>=<replaceable>value</replaceable>
</para>
</parameter>
</syntax>
<description>
<para>Sends a SIP Notify event.</para>
<para>All parameters for this event must be specified in the body of this request
via multiple <literal>Variable: name=value</literal> sequences.</para>
</description>
</manager>
<manager name="SIPpeerstatus" language="en_US">
<synopsis>
Show the status of one or all of the sip peers.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Peer" required="false">
<para>The peer name you want to check.</para>
</parameter>
</syntax>
<description>
<para>Retrieves the status of one or all of the sip peers. If no peer name is specified, status
for all of the sip peers will be retrieved.</para>
</description>
</manager>
<info name="MessageFromInfo" language="en_US" tech="SIP">
<para>The <literal>from</literal> parameter can be a configured peer name
or in the form of "display-name" &lt;URI&gt;.</para>
</info>
<info name="MessageToInfo" language="en_US" tech="SIP">
<para>Specifying a prefix of <literal>sip:</literal> will send the
message as a SIP MESSAGE request.</para>
</info>
<managerEvent language="en_US" name="SIPQualifyPeerDone">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when SIPQualifyPeer has finished qualifying the specified peer.</synopsis>
<syntax>
<parameter name="Peer">
<para>The name of the peer.</para>
</parameter>
<parameter name="ActionID">
<para>This is only included if an ActionID Header was sent with the action request, in which case it will be that ActionID.</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">SIPqualifypeer</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SessionTimeout">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a SIP session times out.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Source">
<para>The source of the session timeout.</para>
<enumlist>
<enum name="RTPTimeout" />
<enum name="SIPSessionTimer" />
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MCID">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Published when a malicious call ID request arrives.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="MCallerIDNumValid">
</parameter>
<parameter name="MCallerIDNum">
</parameter>
<parameter name="MCallerIDton">
</parameter>
<parameter name="MCallerIDNumPlan">
</parameter>
<parameter name="MCallerIDNumPres">
</parameter>
<parameter name="MCallerIDNameValid">
</parameter>
<parameter name="MCallerIDName">
</parameter>
<parameter name="MCallerIDNameCharSet">
</parameter>
<parameter name="MCallerIDNamePres">
</parameter>
<parameter name="MCallerIDSubaddr">
</parameter>
<parameter name="MCallerIDSubaddrType">
</parameter>
<parameter name="MCallerIDSubaddrOdd">
</parameter>
<parameter name="MCallerIDPres">
</parameter>
<parameter name="MConnectedIDNumValid">
</parameter>
<parameter name="MConnectedIDNum">
</parameter>
<parameter name="MConnectedIDton">
</parameter>
<parameter name="MConnectedIDNumPlan">
</parameter>
<parameter name="MConnectedIDNumPres">
</parameter>
<parameter name="MConnectedIDNameValid">
</parameter>
<parameter name="MConnectedIDName">
</parameter>
<parameter name="MConnectedIDNameCharSet">
</parameter>
<parameter name="MConnectedIDNamePres">
</parameter>
<parameter name="MConnectedIDSubaddr">
</parameter>
<parameter name="MConnectedIDSubaddrType">
</parameter>
<parameter name="MConnectedIDSubaddrOdd">
</parameter>
<parameter name="MConnectedIDPres">
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<application name="IAX2Provision" language="en_US">
<synopsis>
Provision a calling IAXy with a given template.
</synopsis>
<syntax>
<parameter name="template">
<para>If not specified, defaults to <literal>default</literal>.</para>
</parameter>
</syntax>
<description>
<para>Provisions the calling IAXy (assuming the calling entity is in fact an IAXy) with the
given <replaceable>template</replaceable>. Returns <literal>-1</literal> on error
or <literal>0</literal> on success.</para>
</description>
</application>
<function name="IAXPEER" language="en_US">
<synopsis>
Gets IAX peer information.
</synopsis>
<syntax>
<parameter name="peername" required="true">
<enumlist>
<enum name="CURRENTCHANNEL">
<para>If <replaceable>peername</replaceable> is specified to this value, return the IP address of the
endpoint of the current channel</para>
</enum>
</enumlist>
</parameter>
<parameter name="item">
<para>If <replaceable>peername</replaceable> is specified, valid items are:</para>
<enumlist>
<enum name="ip">
<para>(default) The IP address.</para>
</enum>
<enum name="status">
<para>The peer's status (if <literal>qualify=yes</literal>)</para>
</enum>
<enum name="mailbox">
<para>The configured mailbox.</para>
</enum>
<enum name="context">
<para>The configured context.</para>
</enum>
<enum name="expire">
<para>The epoch time of the next expire.</para>
</enum>
<enum name="dynamic">
<para>Is it dynamic? (yes/no).</para>
</enum>
<enum name="callerid_name">
<para>The configured Caller ID name.</para>
</enum>
<enum name="callerid_num">
<para>The configured Caller ID number.</para>
</enum>
<enum name="codecs">
<para>The configured codecs.</para>
</enum>
<enum name="codec[x]">
<para>Preferred codec index number <replaceable>x</replaceable> (beginning
with <literal>0</literal>)</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Gets information associated with the specified IAX2 peer.</para>
</description>
<see-also>
<ref type="function">SIPPEER</ref>
</see-also>
</function>
<function name="IAXVAR" language="en_US">
<synopsis>
Sets or retrieves a remote variable.
</synopsis>
<syntax>
<parameter name="varname" required="true" />
</syntax>
<description>
<para>Gets or sets a variable that is sent to a remote IAX2 peer during call setup.</para>
</description>
</function>
<info name="CHANNEL" language="en_US" tech="IAX">
<enumlist>
<enum name="osptoken">
<para>R/O Get the peer's osptoken.</para>
</enum>
<enum name="peerip">
<para>R/O Get the peer's ip address.</para>
</enum>
<enum name="peername">
<para>R/O Get the peer's username.</para>
</enum>
<enum name="secure_signaling">
<para>R/O Get the if the IAX channel is secured.</para>
</enum>
<enum name="secure_media">
<para>R/O Get the if the IAX channel is secured.</para>
</enum>
</enumlist>
</info>
<manager name="IAXpeers" language="en_US">
<synopsis>
List IAX peers.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
</description>
</manager>
<manager name="IAXpeerlist" language="en_US">
<synopsis>
List IAX Peers.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>List all the IAX peers.</para>
</description>
</manager>
<manager name="IAXnetstats" language="en_US">
<synopsis>
Show IAX Netstats.
</synopsis>
<syntax />
<description>
<para>Show IAX channels network statistics.</para>
</description>
</manager>
<manager name="IAXregistry" language="en_US">
<synopsis>
Show IAX registrations.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Show IAX registrations.</para>
</description>
</manager>
<configInfo name="chan_motif" language="en_US">
<synopsis>Jingle Channel Driver</synopsis>
<description>
<para>
<emphasis>Transports</emphasis>
</para>
<para>There are three different transports and protocol derivatives
supported by <literal>chan_motif</literal>. They are in order of
preference: Jingle using ICE-UDP, Google Jingle, and Google-V1.</para>
<para>Jingle as defined in XEP-0166 supports the widest range of
features. It is referred to as <literal>ice-udp</literal>. This is
the specification that Jingle clients implement.</para>
<para>Google Jingle follows the Jingle specification for signaling
but uses a custom transport for media. It is supported by the
Google Talk Plug-in in Gmail and by some other Jingle clients. It
is referred to as <literal>google</literal> in this file.</para>
<para>Google-V1 is the original Google Talk signaling protocol
which uses an initial preliminary version of Jingle. It also uses
the same custom transport as Google Jingle for media. It is
supported by Google Voice, some other Jingle clients, and the
Windows Google Talk client. It is referred to as <literal>google-v1</literal>
in this file.</para>
<para>Incoming sessions will automatically switch to the correct
transport once it has been determined.</para>
<para>Outgoing sessions are capable of determining if the target
is capable of Jingle or a Google transport if the target is in the
roster. Unfortunately it is not possible to differentiate between
a Google Jingle or Google-V1 capable resource until a session
initiate attempt occurs. If a resource is determined to use a
Google transport it will initially use Google Jingle but will fall
back to Google-V1 if required.</para>
<para>If an outgoing session attempt fails due to failure to
support the given transport <literal>chan_motif</literal> will
fall back in preference order listed previously until all
transports have been exhausted.</para>
<para>
<emphasis>Dialing and Resource Selection Strategy</emphasis>
</para>
<para>Placing a call through an endpoint can be accomplished using the
following dial string:</para>
<para>
<literal>Motif/[endpoint name]/[target]</literal>
</para>
<para>When placing an outgoing call through an endpoint the requested
target is searched for in the roster list. If present the first Jingle
or Google Jingle capable resource is specifically targeted. Since the
capabilities of the resource are known the outgoing session initiation
will disregard the configured transport and use the determined one.</para>
<para>If the target is not found in the roster the target will be used
as-is and a session will be initiated using the transport specified
in this configuration file. If no transport has been specified the
endpoint defaults to <literal>ice-udp</literal>.</para>
<para>
<emphasis>Video Support</emphasis>
</para>
<para>Support for video does not need to be explicitly enabled.
Configuring any video codec on your endpoint will automatically enable
it.</para>
<para>
<emphasis>DTMF</emphasis>
</para>
<para>The only supported method for DTMF is RFC2833. This is always
enabled on audio streams and negotiated if possible.</para>
<para>
<emphasis>Incoming Calls</emphasis>
</para>
<para>Incoming calls will first look for the extension matching the
name of the endpoint in the configured context. If no such extension
exists the call will automatically fall back to the <literal>s</literal> extension.</para>
<para>
<emphasis>CallerID</emphasis>
</para>
<para>The incoming caller id number is populated with the username of
the caller and the name is populated with the full identity of the
caller. If you would like to perform authentication or filtering
of incoming calls it is recommended that you use these fields to do so.</para>
<para>Outgoing caller id can <emphasis>not</emphasis> be set.</para>
<warning>
<para>Multiple endpoints using the
same connection is <emphasis>NOT</emphasis> supported. Doing so
may result in broken calls.</para>
</warning>
</description>
<configFile name="motif.conf">
<configObject name="endpoint">
<synopsis>The configuration for an endpoint.</synopsis>
<configOption name="context">
<synopsis>Default dialplan context that incoming sessions will be routed to</synopsis>
</configOption>
<configOption name="callgroup">
<synopsis>A callgroup to assign to this endpoint.</synopsis>
</configOption>
<configOption name="pickupgroup">
<synopsis>A pickup group to assign to this endpoint.</synopsis>
</configOption>
<configOption name="language">
<synopsis>The default language for this endpoint.</synopsis>
</configOption>
<configOption name="musicclass">
<synopsis>Default music on hold class for this endpoint.</synopsis>
</configOption>
<configOption name="parkinglot">
<synopsis>Default parking lot for this endpoint.</synopsis>
</configOption>
<configOption name="accountcode">
<synopsis>Accout code for CDR purposes</synopsis>
</configOption>
<configOption name="allow">
<synopsis>Codecs to allow</synopsis>
</configOption>
<configOption name="disallow">
<synopsis>Codecs to disallow</synopsis>
</configOption>
<configOption name="connection">
<synopsis>Connection to accept traffic on and on which to send traffic out</synopsis>
</configOption>
<configOption name="transport">
<synopsis>The transport to use for the endpoint.</synopsis>
<description>
<para>The default outbound transport for this endpoint. Inbound
messages are inferred. Allowed transports are <literal>ice-udp</literal>,
<literal>google</literal>, or <literal>google-v1</literal>. Note
that <literal>chan_motif</literal> will fall back to transport
preference order if the transport value chosen here fails.</para>
<enumlist>
<enum name="ice-udp">
<para>The Jingle protocol, as defined in XEP 0166.</para>
</enum>
<enum name="google">
<para>The Google Jingle protocol, which follows the Jingle
specification for signaling but uses a custom transport for
media.</para>
</enum>
<enum name="google-v1">
<para>Google-V1 is the original Google Talk signaling
protocol which uses an initial preliminary version of Jingle.
It also uses the same custom transport as <literal>google</literal> for media.</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="maxicecandidates">
<synopsis>Maximum number of ICE candidates to offer</synopsis>
</configOption>
<configOption name="maxpayloads">
<synopsis>Maximum number of pyaloads to offer</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="SKINNYdevices" language="en_US">
<synopsis>
List SKINNY devices (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists Skinny devices in text format with details on current status.
Devicelist will follow as separate events, followed by a final event called
DevicelistComplete.</para>
</description>
</manager>
<manager name="SKINNYshowdevice" language="en_US">
<synopsis>
Show SKINNY device (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Device" required="true">
<para>The device name you want to check.</para>
</parameter>
</syntax>
<description>
<para>Show one SKINNY device with details on current status.</para>
</description>
</manager>
<manager name="SKINNYlines" language="en_US">
<synopsis>
List SKINNY lines (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists Skinny lines in text format with details on current status.
Linelist will follow as separate events, followed by a final event called
LinelistComplete.</para>
</description>
</manager>
<manager name="SKINNYshowline" language="en_US">
<synopsis>
Show SKINNY line (text format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Line" required="true">
<para>The line name you want to check.</para>
</parameter>
</syntax>
<description>
<para>Show one SKINNY line with details on current status.</para>
</description>
</manager>
<function name="DUNDILOOKUP" language="en_US">
<synopsis>
Do a DUNDi lookup of a phone number.
</synopsis>
<syntax>
<parameter name="number" required="true"/>
<parameter name="context">
<para>If not specified the default will be <literal>e164</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="b">
<para>Bypass the internal DUNDi cache</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This will do a DUNDi lookup of the given phone number.</para>
<para>This function will return the Technology/Resource found in the first result
in the DUNDi lookup. If no results were found, the result will be blank.</para>
</description>
</function>
<function name="DUNDIQUERY" language="en_US">
<synopsis>
Initiate a DUNDi query.
</synopsis>
<syntax>
<parameter name="number" required="true"/>
<parameter name="context">
<para>If not specified the default will be <literal>e164</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="b">
<para>Bypass the internal DUNDi cache</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This will do a DUNDi lookup of the given phone number.</para>
<para>The result of this function will be a numeric ID that can be used to retrieve
the results with the <literal>DUNDIRESULT</literal> function.</para>
</description>
</function>
<function name="DUNDIRESULT" language="en_US">
<synopsis>
Retrieve results from a DUNDIQUERY.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The identifier returned by the <literal>DUNDIQUERY</literal> function.</para>
</parameter>
<parameter name="resultnum">
<optionlist>
<option name="number">
<para>The number of the result that you want to retrieve, this starts at <literal>1</literal></para>
</option>
<option name="getnum">
<para>The total number of results that are available.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This function will retrieve results from a previous use\n"
of the <literal>DUNDIQUERY</literal> function.</para>
</description>
</function>
<application name="AELSub" language="en_US">
<synopsis>
Launch subroutine built with AEL
</synopsis>
<syntax>
<parameter name="routine" required="true">
<para>Named subroutine to execute.</para>
</parameter>
<parameter name="args" required="false" />
</syntax>
<description>
<para>Execute the named subroutine, defined in AEL, from another dialplan
language, such as extensions.conf, Realtime extensions, or Lua.</para>
<para>The purpose of this application is to provide a sane entry point into
AEL subroutines, the implementation of which may change from time to time.</para>
</description>
</application>
<manager name="DialplanExtensionAdd" language="en_US">
<synopsis>
Add an extension to the dialplan
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Context" required="true">
<para>Context where the extension will be created. The context will
be created if it does not already exist.</para>
</parameter>
<parameter name="Extension" required="true">
<para>Name of the extension that will be created (may include callerid match by separating
with '/')</para>
</parameter>
<parameter name="Priority" required="true">
<para>Priority being added to this extension. Must be either <literal>hint</literal> or a
numerical value.</para>
</parameter>
<parameter name="Application" required="true">
<para>The application to use for this extension at the requested priority</para>
</parameter>
<parameter name="ApplicationData" required="false">
<para>Arguments to the application.</para>
</parameter>
<parameter name="Replace" required="false">
<para>If set to 'yes', '1', 'true' or any of the other values we evaluate as true, then
if an extension already exists at the requested context, extension, and priority it will
be overwritten. Otherwise, the existing extension will remain and the action will fail.
</para>
</parameter>
</syntax>
</manager>
<manager name="DialplanExtensionRemove" language="en_US">
<synopsis>
Remove an extension from the dialplan
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Context" required="true">
<para>Context of the extension being removed</para>
</parameter>
<parameter name="Extension" required="true">
<para>Name of the extension being removed (may include callerid match by separating with '/')</para>
</parameter>
<parameter name="Priority" required="false">
<para>If provided, only remove this priority from the extension instead of all
priorities in the extension.</para>
</parameter>
</syntax>
</manager>
<application name="Originate" language="en_US">
<synopsis>
Originate a call.
</synopsis>
<syntax>
<parameter name="tech_data" required="true">
<para>Channel technology and data for creating the outbound channel.
For example, SIP/1234.</para>
</parameter>
<parameter name="type" required="true">
<para>This should be <literal>app</literal> or <literal>exten</literal>, depending on whether the outbound channel should be connected to an application or extension.</para>
</parameter>
<parameter name="arg1" required="true">
<para>If the type is <literal>app</literal>, then this is the application name. If the type is <literal>exten</literal>, then this is the context that the channel will be sent to.</para>
</parameter>
<parameter name="arg2" required="false">
<para>If the type is <literal>app</literal>, then this is the data passed as arguments to the application. If the type is <literal>exten</literal>, then this is the extension that the channel will be sent to.</para>
</parameter>
<parameter name="arg3" required="false">
<para>If the type is <literal>exten</literal>, then this is the priority that the channel is sent to. If the type is <literal>app</literal>, then this parameter is ignored.</para>
</parameter>
<parameter name="timeout" required="false">
<para>Timeout in seconds. Default is 30 seconds.</para>
</parameter>
</syntax>
<description>
<para>This application originates an outbound call and connects it to a specified extension or application. This application will block until the outgoing call fails or gets answered. At that point, this application will exit with the status variable set and dialplan processing will continue.</para>
<para>This application sets the following channel variable before exiting:</para>
<variablelist>
<variable name="ORIGINATE_STATUS">
<para>This indicates the result of the call origination.</para>
<value name="FAILED"/>
<value name="SUCCESS"/>
<value name="BUSY"/>
<value name="CONGESTION"/>
<value name="HANGUP"/>
<value name="RINGING"/>
<value name="UNKNOWN">
In practice, you should never see this value. Please report it to the issue tracker if you ever see it.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="SpeechCreate" language="en_US">
<synopsis>
Create a Speech Structure.
</synopsis>
<syntax>
<parameter name="engine_name" required="true" />
</syntax>
<description>
<para>This application creates information to be used by all the other applications.
It must be called before doing any speech recognition activities such as activating a grammar.
It takes the engine name to use as the argument, if not specified the default engine will be used.</para>
<para>Sets the ERROR channel variable to 1 if the engine cannot be used.</para>
</description>
</application>
<application name="SpeechActivateGrammar" language="en_US">
<synopsis>
Activate a grammar.
</synopsis>
<syntax>
<parameter name="grammar_name" required="true" />
</syntax>
<description>
<para>This activates the specified grammar to be recognized by the engine.
A grammar tells the speech recognition engine what to recognize, and how to portray it back to you
in the dialplan. The grammar name is the only argument to this application.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechStart" language="en_US">
<synopsis>
Start recognizing voice in the audio stream.
</synopsis>
<syntax />
<description>
<para>Tell the speech recognition engine that it should start trying to get results from audio being
fed to it.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechBackground" language="en_US">
<synopsis>
Play a sound file and wait for speech to be recognized.
</synopsis>
<syntax>
<parameter name="sound_file" required="true" />
<parameter name="timeout">
<para>Timeout integer in seconds. Note the timeout will only start
once the sound file has stopped playing.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="n">
<para>Don't answer the channel if it has not already been answered.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application plays a sound file and waits for the person to speak. Once they start speaking playback
of the file stops, and silence is heard. Once they stop talking the processing sound is played to indicate
the speech recognition engine is working. Once results are available the application returns and results
(score and text) are available using dialplan functions.</para>
<para>The first text and score are ${SPEECH_TEXT(0)} AND ${SPEECH_SCORE(0)} while the second are ${SPEECH_TEXT(1)}
and ${SPEECH_SCORE(1)}.</para>
<para>The first argument is the sound file and the second is the timeout integer in seconds.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechDeactivateGrammar" language="en_US">
<synopsis>
Deactivate a grammar.
</synopsis>
<syntax>
<parameter name="grammar_name" required="true">
<para>The grammar name to deactivate</para>
</parameter>
</syntax>
<description>
<para>This deactivates the specified grammar so that it is no longer recognized.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechProcessingSound" language="en_US">
<synopsis>
Change background processing sound.
</synopsis>
<syntax>
<parameter name="sound_file" required="true" />
</syntax>
<description>
<para>This changes the processing sound that SpeechBackground plays back when the speech recognition engine is
processing and working to get results.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechDestroy" language="en_US">
<synopsis>
End speech recognition.
</synopsis>
<syntax />
<description>
<para>This destroys the information used by all the other speech recognition applications.
If you call this application but end up wanting to recognize more speech, you must call SpeechCreate()
again before calling any other application.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechLoadGrammar" language="en_US">
<synopsis>
Load a grammar.
</synopsis>
<syntax>
<parameter name="grammar_name" required="true" />
<parameter name="path" required="true" />
</syntax>
<description>
<para>Load a grammar only on the channel, not globally.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<application name="SpeechUnloadGrammar" language="en_US">
<synopsis>
Unload a grammar.
</synopsis>
<syntax>
<parameter name="grammar_name" required="true" />
</syntax>
<description>
<para>Unload a grammar.</para>
<para>Hangs up the channel on failure. If this is not desired, use TryExec.</para>
</description>
</application>
<function name="SPEECH_SCORE" language="en_US">
<synopsis>
Gets the confidence score of a result.
</synopsis>
<syntax argsep="/">
<parameter name="nbest_number" />
<parameter name="result_number" required="true" />
</syntax>
<description>
<para>Gets the confidence score of a result.</para>
</description>
</function>
<function name="SPEECH_TEXT" language="en_US">
<synopsis>
Gets the recognized text of a result.
</synopsis>
<syntax argsep="/">
<parameter name="nbest_number" />
<parameter name="result_number" required="true" />
</syntax>
<description>
<para>Gets the recognized text of a result.</para>
</description>
</function>
<function name="SPEECH_GRAMMAR" language="en_US">
<synopsis>
Gets the matched grammar of a result if available.
</synopsis>
<syntax argsep="/">
<parameter name="nbest_number" />
<parameter name="result_number" required="true" />
</syntax>
<description>
<para>Gets the matched grammar of a result if available.</para>
</description>
</function>
<function name="SPEECH_ENGINE" language="en_US">
<synopsis>
Get or change a speech engine specific attribute.
</synopsis>
<syntax>
<parameter name="name" required="true" />
</syntax>
<description>
<para>Changes a speech engine specific attribute.</para>
</description>
</function>
<function name="SPEECH_RESULTS_TYPE" language="en_US">
<synopsis>
Sets the type of results that will be returned.
</synopsis>
<syntax />
<description>
<para>Sets the type of results that will be returned. Valid options are normal or nbest.</para>
</description>
</function>
<function name="SPEECH" language="en_US">
<synopsis>
Gets information about speech recognition results.
</synopsis>
<syntax>
<parameter name="argument" required="true">
<enumlist>
<enum name="status">
<para>Returns <literal>1</literal> upon speech object existing,
or <literal>0</literal> if not</para>
</enum>
<enum name="spoke">
<para>Returns <literal>1</literal> if spoker spoke,
or <literal>0</literal> if not</para>
</enum>
<enum name="results">
<para>Returns number of results that were recognized.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Gets information about speech recognition results.</para>
</description>
</function>
<application name="PlayTones" language="en_US">
<synopsis>
Play a tone list.
</synopsis>
<syntax>
<parameter name="arg" required="true">
<para>Arg is either the tone name defined in the <filename>indications.conf</filename>
configuration file, or a directly specified list of frequencies and durations.</para>
</parameter>
</syntax>
<description>
<para>Plays a tone list. Execution will continue with the next step in the dialplan
immediately while the tones continue to play.</para>
<para>See the sample <filename>indications.conf</filename> for a description of the
specification of a tonelist.</para>
</description>
<see-also>
<ref type="application">StopPlayTones</ref>
</see-also>
</application>
<application name="StopPlayTones" language="en_US">
<synopsis>
Stop playing a tone list.
</synopsis>
<syntax />
<description>
<para>Stop playing a tone list, initiated by PlayTones().</para>
</description>
<see-also>
<ref type="application">PlayTones</ref>
</see-also>
</application>
<application name="WaitForSilence" language="en_US">
<synopsis>
Waits for a specified amount of silence.
</synopsis>
<syntax>
<parameter name="silencerequired">
<para>If not specified, defaults to <literal>1000</literal> milliseconds.</para>
</parameter>
<parameter name="iterations">
<para>If not specified, defaults to <literal>1</literal>.</para>
</parameter>
<parameter name="timeout">
<para>Is specified only to avoid an infinite loop in cases where silence is never achieved.</para>
</parameter>
</syntax>
<description>
<para>Waits for up to <replaceable>silencerequired</replaceable> milliseconds of silence,
<replaceable>iterations</replaceable> times. An optional <replaceable>timeout</replaceable>
specified the number of seconds to return after, even if we do not receive the specified amount of silence.
Use <replaceable>timeout</replaceable> with caution, as it may defeat the purpose of this application, which
is to wait indefinitely until silence is detected on the line. This is particularly useful for reverse-911-type
call broadcast applications where you need to wait for an answering machine to complete its spiel before
playing a message.</para>
<para>Typically you will want to include two or more calls to WaitForSilence when dealing with an answering
machine; first waiting for the spiel to finish, then waiting for the beep, etc.</para>
<para>Examples:</para>
<para>WaitForSilence(500,2) will wait for 1/2 second of silence, twice</para>
<para>WaitForSilence(1000) will wait for 1 second of silence, once</para>
<para>WaitForSilence(300,3,10) will wait for 300ms silence, 3 times, and returns after 10 sec, even if silence
is not detected</para>
<para>Sets the channel variable <variable>WAITSTATUS</variable> to one of these values:</para>
<variablelist>
<variable name="WAITSTATUS">
<value name="SILENCE">
if exited with silence detected.
</value>
<value name="TIMEOUT">
if exited without silence detected after timeout.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">WaitForNoise</ref>
</see-also>
</application>
<application name="WaitForNoise" language="en_US">
<synopsis>
Waits for a specified amount of noise.
</synopsis>
<syntax>
<parameter name="noiserequired">
<para>If not specified, defaults to <literal>1000</literal> milliseconds.</para>
</parameter>
<parameter name="iterations">
<para>If not specified, defaults to <literal>1</literal>.</para>
</parameter>
<parameter name="timeout">
<para>Is specified only to avoid an infinite loop in cases where silence is never achieved.</para>
</parameter>
</syntax>
<description>
<para>Waits for up to <replaceable>noiserequired</replaceable> milliseconds of noise,
<replaceable>iterations</replaceable> times. An optional <replaceable>timeout</replaceable>
specified the number of seconds to return after, even if we do not receive the specified amount of noise.
Use <replaceable>timeout</replaceable> with caution, as it may defeat the purpose of this application, which
is to wait indefinitely until noise is detected on the line.</para>
</description>
<see-also>
<ref type="application">WaitForSilence</ref>
</see-also>
</application>
<application name="Zapateller" language="en_US">
<synopsis>
Block telemarketers with SIT.
</synopsis>
<syntax>
<parameter name="options" required="true">
<para>Comma delimited list of options.</para>
<optionlist>
<option name="answer">
<para>Causes the line to be answered before playing the tone.</para>
</option>
<option name="nocallerid">
<para>Causes Zapateller to only play the tone if there is no
callerid information available.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Generates special information tone to block telemarketers from calling you.</para>
<para>This application will set the following channel variable upon completion:</para>
<variablelist>
<variable name="ZAPATELLERSTATUS">
<para>This will contain the last action accomplished by the
Zapateller application. Possible values include:</para>
<value name="NOTHING" />
<value name="ANSWERED" />
<value name="ZAPPED" />
</variable>
</variablelist>
</description>
</application>
<application name="TestServer" language="en_US">
<synopsis>
Execute Interface Test Server.
</synopsis>
<syntax />
<description>
<para>Perform test server function and write call report. Results stored in
<filename>/var/log/asterisk/testreports/&lt;testid&gt;-server.txt</filename>
</para>
</description>
<see-also>
<ref type="application">TestClient</ref>
</see-also>
</application>
<application name="TestClient" language="en_US">
<synopsis>
Execute Interface Test Client.
</synopsis>
<syntax>
<parameter name="testid" required="true">
<para>An ID to identify this test.</para>
</parameter>
</syntax>
<description>
<para>Executes test client with given <replaceable>testid</replaceable>. Results stored in
<filename>/var/log/asterisk/testreports/&lt;testid&gt;-client.txt</filename>
</para>
</description>
<see-also>
<ref type="application">TestServer</ref>
</see-also>
</application>
<application name="Read" language="en_US">
<synopsis>
Read a variable.
</synopsis>
<syntax>
<parameter name="variable" required="true">
<para>The input digits will be stored in the given <replaceable>variable</replaceable>
name.</para>
</parameter>
<parameter name="filenames" argsep="&amp;">
<argument name="filename" required="true">
<para>file(s) to play before reading digits or tone with option i</para>
</argument>
<argument name="filename2" multiple="true" />
</parameter>
<parameter name="maxdigits">
<para>Maximum acceptable number of digits. Stops reading after
<replaceable>maxdigits</replaceable> have been entered (without
requiring the user to press the <literal>#</literal> key).</para>
<para>Defaults to <literal>0</literal> - no limit - wait for the
user press the <literal>#</literal> key. Any value below
<literal>0</literal> means the same. Max accepted value is
<literal>255</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="s">
<para>to return immediately if the line is not up.</para>
</option>
<option name="i">
<para>to play filename as an indication tone from your
<filename>indications.conf</filename>.</para>
</option>
<option name="n">
<para>to read digits even if the line is not up.</para>
</option>
</optionlist>
</parameter>
<parameter name="attempts">
<para>If greater than <literal>1</literal>, that many
<replaceable>attempts</replaceable> will be made in the
event no data is entered.</para>
</parameter>
<parameter name="timeout">
<para>The number of seconds to wait for a digit response. If greater
than <literal>0</literal>, that value will override the default timeout.
Can be floating point.</para>
</parameter>
</syntax>
<description>
<para>Reads a #-terminated string of digits a certain number of times from the
user in to the given <replaceable>variable</replaceable>.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="READSTATUS">
<para>This is the status of the read operation.</para>
<value name="OK" />
<value name="ERROR" />
<value name="HANGUP" />
<value name="INTERRUPTED" />
<value name="SKIPPED" />
<value name="TIMEOUT" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">SendDTMF</ref>
</see-also>
</application>
<application name="SendDTMF" language="en_US">
<synopsis>
Sends arbitrary DTMF digits
</synopsis>
<syntax>
<parameter name="digits" required="true">
<para>List of digits 0-9,*#,a-d,A-D to send also w for a half second pause,
W for a one second pause, and f or F for a flash-hook if the channel supports
flash-hook.</para>
</parameter>
<parameter name="timeout_ms" required="false">
<para>Amount of time to wait in ms between tones. (defaults to .25s)</para>
</parameter>
<parameter name="duration_ms" required="false">
<para>Duration of each digit</para>
</parameter>
<parameter name="channel" required="false">
<para>Channel where digits will be played</para>
</parameter>
</syntax>
<description>
<para>It will send all digits or terminate if it encounters an error.</para>
</description>
<see-also>
<ref type="application">Read</ref>
</see-also>
</application>
<manager name="PlayDTMF" language="en_US">
<synopsis>
Play DTMF signal on a specific channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to send digit to.</para>
</parameter>
<parameter name="Digit" required="true">
<para>The DTMF digit to play.</para>
</parameter>
<parameter name="Duration" required="false">
<para>The duration, in milliseconds, of the digit to be played.</para>
</parameter>
</syntax>
<description>
<para>Plays a dtmf digit on the specified channel.</para>
</description>
</manager>
<application name="Authenticate" language="en_US">
<synopsis>
Authenticate a user
</synopsis>
<syntax>
<parameter name="password" required="true">
<para>Password the user should know</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="a">
<para>Set the channels' account code to the password that is entered</para>
</option>
<option name="d">
<para>Interpret the given path as database key, not a literal file.</para>
<note>
<para>The value is not used at all in the authentication when using this option.
If the family/key is set to <literal>/pin/100</literal> (value does not matter)
then the password field needs to be set to <literal>/pin</literal> and the pin entered
by the user would be authenticated against <literal>100</literal>.</para>
</note>
</option>
<option name="m">
<para>Interpret the given path as a file which contains a list of account
codes and password hashes delimited with <literal>:</literal>, listed one per line in
the file. When one of the passwords is matched, the channel will have
its account code set to the corresponding account code in the file.</para>
</option>
<option name="r">
<para>Remove the database key upon successful entry (valid with <literal>d</literal> only)</para>
</option>
</optionlist>
</parameter>
<parameter name="maxdigits" required="false">
<para>maximum acceptable number of digits. Stops reading after
maxdigits have been entered (without requiring the user to press the <literal>#</literal> key).
Defaults to 0 - no limit - wait for the user press the <literal>#</literal> key.</para>
</parameter>
<parameter name="prompt" required="false">
<para>Override the agent-pass prompt file.</para>
</parameter>
</syntax>
<description>
<para>This application asks the caller to enter a given password in order to continue dialplan execution.</para>
<para>If the password begins with the <literal>/</literal> character,
it is interpreted as a file which contains a list of valid passwords, listed 1 password per line in the file.</para>
<para>When using a database key, the value associated with the key can be anything.</para>
<para>Users have three attempts to authenticate before the channel is hung up.</para>
</description>
<see-also>
<ref type="application">VMAuthenticate</ref>
<ref type="application">DISA</ref>
</see-also>
</application>
<application name="Playback" language="en_US">
<synopsis>
Play a file.
</synopsis>
<syntax>
<parameter name="filenames" required="true" argsep="&amp;">
<argument name="filename" required="true" />
<argument name="filename2" multiple="true" />
</parameter>
<parameter name="options">
<para>Comma separated list of options</para>
<optionlist>
<option name="skip">
<para>Do not play if not answered</para>
</option>
<option name="noanswer">
<para>Playback without answering, otherwise the channel will
be answered before the sound is played.</para>
<note>
<para>Not all channel types support playing messages while still on hook.</para>
</note>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Plays back given filenames (do not put extension of wav/alaw etc).
The playback command answer the channel if no options are specified.
If the file is non-existant it will fail</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PLAYBACKSTATUS">
<para>The status of the playback attempt as a text string.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
<para>See Also: Background (application) -- for playing sound files that are interruptible</para>
<para>WaitExten (application) -- wait for digits from caller, optionally play music on hold</para>
</description>
<see-also>
<ref type="application">Background</ref>
<ref type="application">WaitExten</ref>
<ref type="application">ControlPlayback</ref>
<ref type="agi">stream file</ref>
<ref type="agi">control stream file</ref>
<ref type="manager">ControlPlayback</ref>
</see-also>
</application>
<application name="Flash" language="en_US">
<synopsis>
Flashes a DAHDI Trunk.
</synopsis>
<syntax />
<description>
<para>Performs a flash on a DAHDI trunk. This can be used to access features
provided on an incoming analogue circuit such as conference and call waiting.
Use with SendDTMF() to perform external transfers.</para>
</description>
<see-also>
<ref type="application">SendDTMF</ref>
</see-also>
</application>
<application name="BridgeWait" language="en_US">
<synopsis>
Put a call into the holding bridge.
</synopsis>
<syntax>
<parameter name="name">
<para>Name of the holding bridge to join. This is a handle for <literal>BridgeWait</literal>
only and does not affect the actual bridges that are created. If not provided,
the reserved name <literal>default</literal> will be used.
</para>
</parameter>
<parameter name="role" required="false">
<para>Defines the channel's purpose for entering the holding bridge. Values are case sensitive.
</para>
<enumlist>
<enum name="participant">
<para>The channel will enter the holding bridge to be placed on hold
until it is removed from the bridge for some reason. (default)</para>
</enum>
<enum name="announcer">
<para>The channel will enter the holding bridge to make announcements
to channels that are currently in the holding bridge. While an
announcer is present, holding for the participants will be
suspended.</para>
</enum>
</enumlist>
</parameter>
<parameter name="options">
<optionlist>
<option name="m">
<argument name="class" required="true" />
<para>The specified MOH class will be used/suggested for
music on hold operations. This option will only be useful for
entertainment modes that use it (m and h).</para>
</option>
<option name="e">
<para>Which entertainment mechanism should be used while on hold
in the holding bridge. Only the first letter is read.</para>
<enumlist>
<enum name="m">
<para>Play music on hold (default)</para>
</enum>
<enum name="r">
<para>Ring without pause</para>
</enum>
<enum name="s">
<para>Generate silent audio</para>
</enum>
<enum name="h">
<para>Put the channel on hold</para>
</enum>
<enum name="n">
<para>No entertainment</para>
</enum>
</enumlist>
</option>
<option name="S">
<argument name="duration" required="true" />
<para>Automatically exit the bridge and return to the PBX after
<emphasis>duration</emphasis> seconds.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application places the incoming channel into a holding bridge.
The channel will then wait in the holding bridge until some event occurs
which removes it from the holding bridge.</para>
<note>
<para>This application will answer calls which haven't already
been answered.</para>
</note>
</description>
</application>
<application name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</synopsis>
<syntax>
<parameter name="file" required="true" argsep=".">
<argument name="filename" required="true">
<para>If <replaceable>filename</replaceable> is an absolute path, uses that path, otherwise
creates the file in the configured monitoring directory from <filename>asterisk.conf.</filename>
</para>
</argument>
<argument name="extension" required="true" />
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Append to the file instead of overwriting it.</para>
</option>
<option name="b">
<para>Only save audio to the file while the channel is bridged.</para>
<note>
<para>If you utilize this option inside a Local channel, you must make sure the Local
channel is not optimized away. To do this, be sure to call your Local channel with the
<literal>/n</literal> option. For example: Dial(Local/start@mycontext/n)</para>
</note>
</option>
<option name="B">
<para>Play a periodic beep while this call is being recorded.</para>
<argument name="interval">
<para>Interval, in seconds. Default is 15.</para>
</argument>
</option>
<option name="v">
<para>Adjust the <emphasis>heard</emphasis> volume by a factor of <replaceable>x</replaceable>
(range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="V">
<para>Adjust the <emphasis>spoken</emphasis> volume by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="W">
<para>Adjust both, <emphasis>heard and spoken</emphasis> volumes by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="r">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>receive</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="t">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>transmit</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="i">
<argument name="chanvar" required="true" />
<para>Stores the MixMonitor's ID on this channel variable.</para>
</option>
<option name="p">
<para>Play a beep on the channel that starts the recording.</para>
</option>
<option name="P">
<para>Play a beep on the channel that stops the recording.</para>
</option>
<option name="m">
<argument name="mailbox" required="true" />
<para>Create a copy of the recording as a voicemail in the indicated <emphasis>mailbox</emphasis>(es)
separated by commas eg. m(1111@default,2222@default,...). Folders can be optionally specified using
the syntax: mailbox@context/folder</para>
</option>
</optionlist>
</parameter>
<parameter name="command">
<para>Will be executed when the recording is over.</para>
<para>Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.</para>
<para>All variables will be evaluated at the time MixMonitor is called.</para>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</parameter>
</syntax>
<description>
<para>Records the audio on the current channel to the specified file.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer or Progress().</para>
<note>
<para>MixMonitor runs as an audiohook.</para>
</note>
<variablelist>
<variable name="MIXMONITOR_FILENAME">
<para>Will contain the filename used to record.</para>
</variable>
</variablelist>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of ANY of the application's
parameters. You risk a command injection attack executing arbitrary commands
if the untrusted strings aren't filtered to remove dangerous characters. See
function <variable>FILTER()</variable>.</para>
</warning>
</description>
<see-also>
<ref type="application">Monitor</ref>
<ref type="application">StopMixMonitor</ref>
<ref type="application">PauseMonitor</ref>
<ref type="application">UnpauseMonitor</ref>
<ref type="function">AUDIOHOOK_INHERIT</ref>
</see-also>
</application>
<application name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor, and free the recording's file handle.
</synopsis>
<syntax>
<parameter name="MixMonitorID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
MixMonitor.</para>
</parameter>
</syntax>
<description>
<para>Stops the audio recording that was started with a call to <literal>MixMonitor()</literal>
on the current channel.</para>
</description>
<see-also>
<ref type="application">MixMonitor</ref>
</see-also>
</application>
<manager name="MixMonitorMute" language="en_US">
<synopsis>
Mute / unMute a Mixmonitor recording.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to mute.</para>
</parameter>
<parameter name="Direction">
<para>Which part of the recording to mute: read, write or both (from channel, to channel or both channels).</para>
</parameter>
<parameter name="State">
<para>Turn mute on or off : 1 to turn on, 0 to turn off.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to mute a MixMonitor recording.</para>
</description>
</manager>
<manager name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
<parameter name="File">
<para>Is the name of the file created in the monitor spool directory.
Defaults to the same name as the channel (with slashes replaced with dashes).
This argument is optional if you specify to record unidirectional audio with
either the r(filename) or t(filename) options in the options field. If
neither MIXMONITOR_FILENAME or this parameter is set, the mixed stream won't
be recorded.</para>
</parameter>
<parameter name="options">
<para>Options that apply to the MixMonitor in the same way as they
would apply if invoked from the MixMonitor application. For a list of
available options, see the documentation for the mixmonitor application. </para>
</parameter>
<parameter name="Command">
<para>Will be executed when the recording is over.
Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.
All variables will be evaluated at the time MixMonitor is called.</para>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</parameter>
</syntax>
<description>
<para>This action records the audio on the current channel to the specified file.</para>
<variablelist>
<variable name="MIXMONITOR_FILENAME">
<para>Will contain the filename used to record the mixed stream.</para>
</variable>
</variablelist>
</description>
</manager>
<manager name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor, and free the recording's file handle.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel monitored.</para>
</parameter>
<parameter name="MixMonitorID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
MixMonitor.</para>
</parameter>
</syntax>
<description>
<para>This action stops the audio recording that was started with the <literal>MixMonitor</literal>
action on the current channel.</para>
</description>
</manager>
<function name="MIXMONITOR" language="en_US">
<synopsis>
Retrieve data pertaining to specific instances of MixMonitor on a channel.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The unique ID of the MixMonitor instance. The unique ID can be retrieved through the channel
variable used as an argument to the <replaceable>i</replaceable> option to MixMonitor.</para>
</parameter>
<parameter name="key" required="true">
<para>The piece of data to retrieve from the MixMonitor.</para>
<enumlist>
<enum name="filename" />
</enumlist>
</parameter>
</syntax>
</function>
<application name="ConfBridge" language="en_US">
<synopsis>
Conference bridge application.
</synopsis>
<syntax>
<parameter name="conference" required="true">
<para>Name of the conference bridge. You are not limited to just
numbers.</para>
</parameter>
<parameter name="bridge_profile">
<para>The bridge profile name from confbridge.conf. When left blank,
a dynamically built bridge profile created by the CONFBRIDGE dialplan
function is searched for on the channel and used. If no dynamic
profile is present, the 'default_bridge' profile found in
confbridge.conf is used. </para>
<para>It is important to note that while user profiles may be unique
for each participant, mixing bridge profiles on a single conference
is _NOT_ recommended and will produce undefined results.</para>
</parameter>
<parameter name="user_profile">
<para>The user profile name from confbridge.conf. When left blank,
a dynamically built user profile created by the CONFBRIDGE dialplan
function is searched for on the channel and used. If no dynamic
profile is present, the 'default_user' profile found in
confbridge.conf is used.</para>
</parameter>
<parameter name="menu">
<para>The name of the DTMF menu in confbridge.conf to be applied to
this channel. When left blank, a dynamically built menu profile
created by the CONFBRIDGE dialplan function is searched for on
the channel and used. If no dynamic profile is present, the
'default_menu' profile found in confbridge.conf is used.</para>
</parameter>
</syntax>
<description>
<para>Enters the user into a specified conference bridge. The user can
exit the conference by hangup or DTMF menu option.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="CONFBRIDGE_RESULT">
<value name="FAILED">The channel encountered an error and could not enter the conference.</value>
<value name="HANGUP">The channel exited the conference by hanging up.</value>
<value name="KICKED">The channel was kicked from the conference.</value>
<value name="ENDMARKED">The channel left the conference as a result of the last marked user leaving.</value>
<value name="DTMF">The channel pressed a DTMF sequence to exit the conference.</value>
<value name="TIMEOUT">The channel reached its configured timeout.</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">ConfBridge</ref>
<ref type="function">CONFBRIDGE</ref>
<ref type="function">CONFBRIDGE_INFO</ref>
</see-also>
</application>
<function name="CONFBRIDGE" language="en_US">
<synopsis>
Set a custom dynamic bridge, user, or menu profile on a channel for the
ConfBridge application using the same options available in confbridge.conf.
</synopsis>
<syntax>
<parameter name="type" required="true">
<para>To what type of conference profile the option applies.</para>
<enumlist>
<enum name="bridge"></enum>
<enum name="menu"></enum>
<enum name="user"></enum>
</enumlist>
</parameter>
<parameter name="option" required="true">
<para>Option refers to a <filename>confbridge.conf</filename> option
that is being set dynamically on this channel, or <literal>clear</literal>
to remove already applied profile options from the channel.</para>
</parameter>
</syntax>
<description>
<para>A custom profile uses the default profile type settings defined in
<filename>confbridge.conf</filename> as defaults if the profile template
is not explicitly specified first.</para>
<para>For <literal>bridge</literal> profiles the default template is <literal>default_bridge</literal>.</para>
<para>For <literal>menu</literal> profiles the default template is <literal>default_menu</literal>.</para>
<para>For <literal>user</literal> profiles the default template is <literal>default_user</literal>.</para>
<para>---- Example 1 ----</para>
<para>In this example the custom user profile set on the channel will
automatically be used by the ConfBridge application.</para>
<para>exten => 1,1,Answer()</para>
<para>; In this example the effect of the following line is</para>
<para>; implied:</para>
<para>; same => n,Set(CONFBRIDGE(user,template)=default_user)</para>
<para>same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)</para>
<para>same => n,Set(CONFBRIDGE(user,startmuted)=yes)</para>
<para>same => n,ConfBridge(1) </para>
<para>---- Example 2 ----</para>
<para>This example shows how to use a predefined user profile in
<filename>confbridge.conf</filename> as a template for a dynamic profile.
Here we make an admin/marked user out of the <literal>my_user</literal>
profile that you define in <filename>confbridge.conf</filename>.</para>
<para>exten => 1,1,Answer()</para>
<para>same => n,Set(CONFBRIDGE(user,template)=my_user)</para>
<para>same => n,Set(CONFBRIDGE(user,admin)=yes)</para>
<para>same => n,Set(CONFBRIDGE(user,marked)=yes)</para>
<para>same => n,ConfBridge(1)</para>
</description>
</function>
<function name="CONFBRIDGE_INFO" language="en_US">
<synopsis>
Get information about a ConfBridge conference.
</synopsis>
<syntax>
<parameter name="type" required="true">
<para>What conference information is requested.</para>
<enumlist>
<enum name="admins">
<para>Get the number of admin users in the conference.</para>
</enum>
<enum name="locked">
<para>Determine if the conference is locked. (0 or 1)</para>
</enum>
<enum name="marked">
<para>Get the number of marked users in the conference.</para>
</enum>
<enum name="muted">
<para>Determine if the conference is muted. (0 or 1)</para>
</enum>
<enum name="parties">
<para>Get the number of users in the conference.</para>
</enum>
</enumlist>
</parameter>
<parameter name="conf" required="true">
<para>The name of the conference being referenced.</para>
</parameter>
</syntax>
<description>
<para>This function returns a non-negative integer for valid conference
names and an empty string for invalid conference names.</para>
</description>
</function>
<manager name="ConfbridgeList" language="en_US">
<synopsis>
List participants in a conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true">
<para>Conference number.</para>
</parameter>
</syntax>
<description>
<para>Lists all users in a particular ConfBridge conference.
ConfbridgeList will follow as separate events, followed by a final event called
ConfbridgeListComplete.</para>
</description>
</manager>
<managerEvent language="en_US" name="ConfbridgeList">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>Raised as part of the ConfbridgeList action response list.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="MarkedUser">
<para>Identifies this user as a marked user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="WaitMarked">
<para>Must this user wait for a marked user to join?</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="EndMarked">
<para>Does this user get kicked after the last marked user leaves?</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="Waiting">
<para>Is this user waiting for a marked user to join?</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="Muted">
<para>The current mute status.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="AnsweredTime">
<para>The number of seconds the channel has been up.</para>
</parameter>
<channel_snapshot/>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="ConfbridgeListRooms" language="en_US">
<synopsis>
List active conferences.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists data about all active conferences.
ConfbridgeListRooms will follow as separate events, followed by a final event called
ConfbridgeListRoomsComplete.</para>
</description>
</manager>
<manager name="ConfbridgeMute" language="en_US">
<synopsis>
Mute a Confbridge user.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
<parameter name="Channel" required="true">
<para>If this parameter is not a complete channel name, the first channel with this prefix will be used.</para>
<para>If this parameter is "all", all channels will be muted.</para>
<para>If this parameter is "participants", all non-admin channels will be muted.</para>
</parameter>
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeUnmute" language="en_US">
<synopsis>
Unmute a Confbridge user.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
<parameter name="Channel" required="true">
<para>If this parameter is not a complete channel name, the first channel with this prefix will be used.</para>
<para>If this parameter is "all", all channels will be unmuted.</para>
<para>If this parameter is "participants", all non-admin channels will be unmuted.</para>
</parameter>
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeKick" language="en_US">
<synopsis>
Kick a Confbridge user.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
<parameter name="Channel" required="true" >
<para>If this parameter is "all", all channels will be kicked from the conference.</para>
<para>If this parameter is "participants", all non-admin channels will be kicked from the conference.</para>
</parameter>
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeLock" language="en_US">
<synopsis>
Lock a Confbridge conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeUnlock" language="en_US">
<synopsis>
Unlock a Confbridge conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeStartRecord" language="en_US">
<synopsis>
Start recording a Confbridge conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
<parameter name="RecordFile" required="false" />
</syntax>
<description>
<para>Start recording a conference. If recording is already present an error will be returned. If RecordFile is not provided, the default record file specified in the conference's bridge profile will be used, if that is not present either a file will automatically be generated in the monitor directory.</para>
</description>
</manager>
<manager name="ConfbridgeStopRecord" language="en_US">
<synopsis>
Stop recording a Confbridge conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="ConfbridgeSetSingleVideoSrc" language="en_US">
<synopsis>
Set a conference user as the single video source distributed to all other participants.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="true" />
<parameter name="Channel" required="true">
<para>If this parameter is not a complete channel name, the first channel with this prefix will be used.</para>
</parameter>
</syntax>
<description>
</description>
</manager>
<application name="Directory" language="en_US">
<synopsis>
Provide directory of voicemail extensions.
</synopsis>
<syntax>
<parameter name="vm-context">
<para>This is the context within voicemail.conf to use for the Directory. If not
specified and <literal>searchcontexts=no</literal> in
<filename>voicemail.conf</filename>, then <literal>default</literal>
will be assumed.</para>
</parameter>
<parameter name="dial-context" required="false">
<para>This is the dialplan context to use when looking for an
extension that the user has selected, or when jumping to the
<literal>o</literal> or <literal>a</literal> extension. If not
specified, the current context will be used.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="e">
<para>In addition to the name, also read the extension number to the
caller before presenting dialing options.</para>
</option>
<option name="f">
<para>Allow the caller to enter the first name of a user in the
directory instead of using the last name. If specified, the
optional number argument will be used for the number of
characters the user should enter.</para>
<argument name="n" required="true" />
</option>
<option name="l">
<para>Allow the caller to enter the last name of a user in the
directory. This is the default. If specified, the
optional number argument will be used for the number of
characters the user should enter.</para>
<argument name="n" required="true" />
</option>
<option name="b">
<para> Allow the caller to enter either the first or the last name
of a user in the directory. If specified, the optional number
argument will be used for the number of characters the user should enter.</para>
<argument name="n" required="true" />
</option>
<option name="a">
<para>Allow the caller to additionally enter an alias for a user in the
directory. This option must be specified in addition to the
<literal>f</literal>, <literal>l</literal>, or <literal>b</literal>
option.</para>
</option>
<option name="m">
<para>Instead of reading each name sequentially and asking for
confirmation, create a menu of up to 8 names.</para>
</option>
<option name="n">
<para>Read digits even if the channel is not answered.</para>
</option>
<option name="p">
<para>Pause for n milliseconds after the digits are typed. This is
helpful for people with cellphones, who are not holding the
receiver to their ear while entering DTMF.</para>
<argument name="n" required="true" />
</option>
</optionlist>
<note>
<para>Only one of the <replaceable>f</replaceable>, <replaceable>l</replaceable>, or <replaceable>b</replaceable>
options may be specified. <emphasis>If more than one is specified</emphasis>, then Directory will act as
if <replaceable>b</replaceable> was specified. The number
of characters for the user to type defaults to <literal>3</literal>.</para>
</note>
</parameter>
</syntax>
<description>
<para>This application will present the calling channel with a directory of extensions from which they can search
by name. The list of names and corresponding extensions is retrieved from the
voicemail configuration file, <filename>voicemail.conf</filename>.</para>
<para>This application will immediately exit if one of the following DTMF digits are
received and the extension to jump to exists:</para>
<para>
<literal>0</literal> - Jump to the 'o' extension, if it exists.</para>
<para>
<literal>*</literal> - Jump to the 'a' extension, if it exists.</para>
<para>This application will set the following channel variable before completion:</para>
<variablelist>
<variable name="DIRECTORY_RESULT">
<para>Reason Directory application exited.</para>
<value name="OPERATOR">User requested operator</value>
<value name="ASSISTANT">User requested assistant</value>
<value name="TIMEOUT">User allowed DTMF wait duration to pass without sending DTMF</value>
<value name="HANGUP">The channel hung up before the application finished</value>
<value name="SELECTED">User selected a user to call from the directory</value>
<value name="USEREXIT">User exited with '#' during selection</value>
<value name="FAILED">The application failed</value>
</variable>
</variablelist>
</description>
</application>
<application name="Page" language="en_US">
<synopsis>
Page series of phones
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<para>Specification of the device(s) to dial. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
available to that particular channel driver.</para>
</argument>
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to dial in parallel</para>
<para>If you need more than one, enter them as Technology2/Resource2&amp;
Technology3/Resource3&amp;.....</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="b" argsep="^">
<para>Before initiating an outgoing call, Gosub to the specified
location using the newly created channel. The Gosub will be
executed for each destination channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="B" argsep="^">
<para>Before initiating the outgoing call(s), Gosub to the specified
location using the current channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="d">
<para>Full duplex audio</para>
</option>
<option name="i">
<para>Ignore attempts to forward the call</para>
</option>
<option name="q">
<para>Quiet, do not play beep to caller</para>
</option>
<option name="r">
<para>Record the page into a file (<literal>CONFBRIDGE(bridge,record_conference)</literal>)</para>
</option>
<option name="s">
<para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
</option>
<option name="A">
<argument name="x" required="true">
<para>The announcement to playback to all devices</para>
</argument>
<para>Play an announcement to all paged participants</para>
</option>
<option name="n">
<para>Do not play announcement to caller (alters <literal>A(x)</literal> behavior)</para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>Specify the length of time that the system will attempt to connect a call.
After this duration, any page calls that have not been answered will be hung up by the
system.</para>
</parameter>
</syntax>
<description>
<para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
and dumps them into a conference bridge as muted participants. The original
caller is dumped into the conference as a speaker and the room is
destroyed when the original caller leaves.</para>
</description>
<see-also>
<ref type="application">ConfBridge</ref>
</see-also>
</application>
<application name="UserEvent" language="en_US">
<synopsis>
Send an arbitrary user-defined event to parties interested in a channel (AMI users and relevant res_stasis applications).
</synopsis>
<syntax>
<parameter name="eventname" required="true" />
<parameter name="body" />
</syntax>
<description>
<para>Sends an arbitrary event to interested parties, with an optional
<replaceable>body</replaceable> representing additional arguments. The
<replaceable>body</replaceable> may be specified as
a <literal>,</literal> delimited list of key:value pairs.</para>
<para>For AMI, each additional argument will be placed on a new line in
the event and the format of the event will be:</para>
<para> Event: UserEvent</para>
<para> UserEvent: &lt;specified event name&gt;</para>
<para> [body]</para>
<para>If no <replaceable>body</replaceable> is specified, only Event and
UserEvent headers will be present.</para>
<para>For res_stasis applications, the event will be provided as a JSON
blob with additional arguments appearing as keys in the object and the
<replaceable>eventname</replaceable> under the
<literal>eventname</literal> key.</para>
</description>
<see-also>
<ref type="manager">UserEvent</ref>
<ref type="managerEvent">UserEvent</ref>
</see-also>
</application>
<application name="Transfer" language="en_US">
<synopsis>
Transfer caller to remote extension.
</synopsis>
<syntax>
<parameter name="dest" required="true" argsep="">
<argument name="Tech/" />
<argument name="destination" required="true" />
</parameter>
</syntax>
<description>
<para>Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming call with the same channel technology will be transferred.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.</para>
<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable>
channel variable:</para>
<variablelist>
<variable name="TRANSFERSTATUS">
<value name="SUCCESS">
Transfer succeeded.
</value>
<value name="FAILURE">
Transfer failed.
</value>
<value name="UNSUPPORTED">
Transfer unsupported by channel driver.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="Exec" language="en_US">
<synopsis>
Executes dialplan application.
</synopsis>
<syntax>
<parameter name="appname" required="true" hasparams="true">
<para>Application name and arguments of the dialplan application to execute.</para>
<argument name="arguments" required="true" />
</parameter>
</syntax>
<description>
<para>Allows an arbitrary application to be invoked even when not
hard coded into the dialplan. If the underlying application
terminates the dialplan, or if the application cannot be found,
Exec will terminate the dialplan.</para>
<para>To invoke external applications, see the application System.
If you would like to catch any error instead, see TryExec.</para>
</description>
</application>
<application name="TryExec" language="en_US">
<synopsis>
Executes dialplan application, always returning.
</synopsis>
<syntax>
<parameter name="appname" required="true" hasparams="true">
<argument name="arguments" required="true" />
</parameter>
</syntax>
<description>
<para>Allows an arbitrary application to be invoked even when not
hard coded into the dialplan. To invoke external applications
see the application System. Always returns to the dialplan.
The channel variable TRYSTATUS will be set to one of:
</para>
<variablelist>
<variable name="TRYSTATUS">
<value name="SUCCESS">
If the application returned zero.
</value>
<value name="FAILED">
If the application returned non-zero.
</value>
<value name="NOAPP">
If the application was not found or was not specified.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="ExecIf" language="en_US">
<synopsis>
Executes dialplan application, conditionally.
</synopsis>
<syntax argsep="?">
<parameter name="expression" required="true" />
<parameter name="execapp" required="true" argsep=":">
<argument name="appiftrue" required="true" hasparams="true">
<argument name="args" required="true" />
</argument>
<argument name="appiffalse" required="false" hasparams="true">
<argument name="args" required="true" />
</argument>
</parameter>
</syntax>
<description>
<para>If <replaceable>expr</replaceable> is true, execute and return the
result of <replaceable>appiftrue(args)</replaceable>.</para>
<para>If <replaceable>expr</replaceable> is true, but <replaceable>appiftrue</replaceable> is not found,
then the application will return a non-zero value.</para>
</description>
</application>
<application name="ControlPlayback" language="en_US">
<synopsis>
Play a file with fast forward and rewind.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="skipms">
<para>This is number of milliseconds to skip when rewinding or
fast-forwarding.</para>
</parameter>
<parameter name="ff">
<para>Fast-forward when this DTMF digit is received. (defaults to <literal>#</literal>)</para>
</parameter>
<parameter name="rew">
<para>Rewind when this DTMF digit is received. (defaults to <literal>*</literal>)</para>
</parameter>
<parameter name="stop">
<para>Stop playback when this DTMF digit is received.</para>
</parameter>
<parameter name="pause">
<para>Pause playback when this DTMF digit is received.</para>
</parameter>
<parameter name="restart">
<para>Restart playback when this DTMF digit is received.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="o">
<argument name="time" required="true">
<para>Start at <replaceable>time</replaceable> ms from the
beginning of the file.</para>
</argument>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application will play back the given <replaceable>filename</replaceable>.</para>
<para>It sets the following channel variables upon completion:</para>
<variablelist>
<variable name="CPLAYBACKSTATUS">
<para>Contains the status of the attempt as a text string</para>
<value name="SUCCESS" />
<value name="USERSTOPPED" />
<value name="REMOTESTOPPED" />
<value name="ERROR" />
</variable>
<variable name="CPLAYBACKOFFSET">
<para>Contains the offset in ms into the file where playback
was at when it stopped. <literal>-1</literal> is end of file.</para>
</variable>
<variable name="CPLAYBACKSTOPKEY">
<para>If the playback is stopped by the user this variable contains
the key that was pressed.</para>
</variable>
</variablelist>
</description>
</application>
<manager name="ControlPlayback" language="en_US">
<synopsis>
Control the playback of a file being played to a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel that currently has a file being played back to it.</para>
</parameter>
<parameter name="Control" required="true">
<enumlist>
<enum name="stop">
<para>Stop the playback operation.</para>
</enum>
<enum name="forward">
<para>Move the current position in the media forward. The amount
of time that the stream moves forward is determined by the
<replaceable>skipms</replaceable> value passed to the application
that initiated the playback.</para>
<note>
<para>The default skipms value is <literal>3000</literal> ms.</para>
</note>
</enum>
<enum name="reverse">
<para>Move the current position in the media backward. The amount
of time that the stream moves backward is determined by the
<replaceable>skipms</replaceable> value passed to the application
that initiated the playback.</para>
<note>
<para>The default skipms value is <literal>3000</literal> ms.</para>
</note>
</enum>
<enum name="pause">
<para>Pause/unpause the playback operation, if supported.
If not supported, stop the playback.</para>
</enum>
<enum name="restart">
<para>Restart the playback operation, if supported.
If not supported, stop the playback.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Control the operation of a media file being played back to a channel.
Note that this AMI action does not initiate playback of media to channel, but
rather controls the operation of a media operation that was already initiated
on the channel.</para>
<note>
<para>The <literal>pause</literal> and <literal>restart</literal>
<replaceable>Control</replaceable> options will stop a playback
operation if that operation was not initiated from the
<replaceable>ControlPlayback</replaceable> application or the
<replaceable>control stream file</replaceable> AGI command.</para>
</note>
</description>
<see-also>
<ref type="application">Playback</ref>
<ref type="application">ControlPlayback</ref>
<ref type="agi">stream file</ref>
<ref type="agi">control stream file</ref>
</see-also>
</manager>
<application name="Dial" language="en_US">
<synopsis>
Attempt to connect to another device or endpoint and bridge the call.
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<para>Specification of the device(s) to dial. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable>
represents a resource available to that particular channel driver.</para>
</argument>
<argument name="Technology2/Resource2" required="false" multiple="true">
<para>Optional extra devices to dial in parallel</para>
<para>If you need more than one enter them as
Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
</argument>
</parameter>
<parameter name="timeout" required="false">
<para>Specifies the number of seconds we attempt to dial the specified devices.</para>
<para>If not specified, this defaults to 136 years.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="A">
<argument name="x" required="true">
<para>The file to play to the called party</para>
</argument>
<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
</option>
<option name="a">
<para>Immediately answer the calling channel when the called channel answers in
all cases. Normally, the calling channel is answered when the called channel
answers, but when options such as <literal>A()</literal> and
<literal>M()</literal> are used, the calling channel is
not answered until all actions on the called channel (such as playing an
announcement) are completed. This option can be used to answer the calling
channel before doing anything on the called channel. You will rarely need to use
this option, the default behavior is adequate in most cases.</para>
</option>
<option name="b" argsep="^">
<para>Before initiating an outgoing call, <literal>Gosub</literal> to the specified
location using the newly created channel. The <literal>Gosub</literal> will be
executed for each destination channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="B" argsep="^">
<para>Before initiating the outgoing call(s), <literal>Gosub</literal> to the
specified location using the current channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="C">
<para>Reset the call detail record (CDR) for this call.</para>
</option>
<option name="c">
<para>If the Dial() application cancels this call, always set
<variable>HANGUPCAUSE</variable> to 'answered elsewhere'</para>
</option>
<option name="d">
<para>Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
if it exists.</para>
<note>
<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
connected. If you wish to use this option with these phones, you
can use the <literal>Answer</literal> application before dialing.</para>
</note>
</option>
<option name="D" argsep=":">
<argument name="called" />
<argument name="calling" />
<argument name="progress" />
<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
party has answered, but before the call gets bridged. The
<replaceable>called</replaceable> DTMF string is sent to the called party, and the
<replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
<para>See <literal>SendDTMF</literal> for valid digits.</para>
</option>
<option name="e">
<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
</option>
<option name="f">
<argument name="x" required="false" />
<para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
deflection to the dialplan extension of this <literal>Dial()</literal> using a dialplan <literal>hint</literal>.
For example, some PSTNs do not allow CallerID to be set to anything
other than the numbers assigned to you.
If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
</option>
<option name="F" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
</option>
<option name="F">
<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
and <emphasis>start</emphasis> execution at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
<note>
<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
</note>
</option>
<option name="g">
<para>Proceed with dialplan execution at the next priority in the current extension if the
destination channel hangs up.</para>
</option>
<option name="G" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>If the call is answered, transfer the calling party to
the specified <replaceable>priority</replaceable> and the called party to the specified
<replaceable>priority</replaceable> plus one.</para>
<note>
<para>You cannot use any additional action post answer options in conjunction with this option.</para>
</note>
</option>
<option name="h">
<para>Allow the called party to hang up by sending the DTMF sequence
defined for disconnect in <filename>features.conf</filename>.</para>
</option>
<option name="H">
<para>Allow the calling party to hang up by sending the DTMF sequence
defined for disconnect in <filename>features.conf</filename>.</para>
<note>
<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
connected. If you wish to allow DTMF disconnect before the dialed
party answers with these phones, you can use the <literal>Answer</literal>
application before dialing.</para>
</note>
</option>
<option name="i">
<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
</option>
<option name="I">
<para>Asterisk will ignore any connected line update requests or any redirecting party
update requests it may receive on this dial attempt.</para>
</option>
<option name="k">
<para>Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="K">
<para>Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="L" argsep=":">
<argument name="x" required="true">
<para>Maximum call time, in milliseconds</para>
</argument>
<argument name="y">
<para>Warning time, in milliseconds</para>
</argument>
<argument name="z">
<para>Repeat time, in milliseconds</para>
</argument>
<para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
<para>This option is affected by the following variables:</para>
<variablelist>
<variable name="LIMIT_PLAYAUDIO_CALLER">
<value name="yes" default="true" />
<value name="no" />
<para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
</variable>
<variable name="LIMIT_PLAYAUDIO_CALLEE">
<value name="yes" />
<value name="no" default="true"/>
<para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
</variable>
<variable name="LIMIT_TIMEOUT_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_CONNECT_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_WARNING_FILE">
<value name="filename"/>
<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
</variable>
</variablelist>
</option>
<option name="m">
<argument name="class" required="false"/>
<para>Provide hold music to the calling party until a requested
channel answers. A specific music on hold <replaceable>class</replaceable>
(as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
</option>
<option name="M" argsep="^">
<argument name="macro" required="true">
<para>Name of the macro that should be executed.</para>
</argument>
<argument name="arg" multiple="true">
<para>Macro arguments</para>
</argument>
<para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
before connecting to the calling channel. Arguments can be specified to the Macro
using <literal>^</literal> as a delimiter. The macro can set the variable
<variable>MACRO_RESULT</variable> to specify the following actions after the macro is
finished executing:</para>
<variablelist>
<variable name="MACRO_RESULT">
<para>If set, this action will be taken after the macro finished executing.</para>
<value name="ABORT">
Hangup both legs of the call
</value>
<value name="CONGESTION">
Behave as if line congestion was encountered
</value>
<value name="BUSY">
Behave as if a busy signal was encountered
</value>
<value name="CONTINUE">
Hangup the called party and allow the calling party to continue dialplan execution at the next priority
</value>
<value name="GOTO:[[&lt;context&gt;^]&lt;exten&gt;^]&lt;priority&gt;">
Transfer the call to the specified destination.
</value>
</variable>
</variablelist>
<note>
<para>You cannot use any additional action post answer options in conjunction
with this option. Also, pbx services are run on the peer (called) channel,
so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this macro.</para>
</note>
<warning>
<para>Be aware of the limitations that macros have, specifically with regards to use of
the <literal>WaitExten</literal> application. For more information, see the documentation for
<literal>Macro()</literal>.</para>
</warning>
</option>
<option name="n">
<argument name="delete">
<para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
the recorded introduction will not be deleted if the caller hangs up while the remote party has not
yet answered.</para>
<para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
always be deleted.</para>
</argument>
<para>This option is a modifier for the call screening/privacy mode. (See the
<literal>p</literal> and <literal>P</literal> options.) It specifies
that no introductions are to be saved in the <directory>priv-callerintros</directory>
directory.</para>
</option>
<option name="N">
<para>This option is a modifier for the call screening/privacy mode. It specifies
that if CallerID is present, do not screen the call.</para>
</option>
<option name="o">
<argument name="x" required="false" />
<para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
<emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
This was the behavior of Asterisk 1.0 and earlier.
If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
Note that <literal>o(${CALLERID(all)})</literal> is similar to option <literal>o</literal> without the parameter.</para>
</option>
<option name="O">
<argument name="mode">
<para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
the originator hanging up will cause the phone to ring back immediately.</para>
<para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
flashes the trunk, it will ring their phone back.</para>
</argument>
<para>Enables <emphasis>operator services</emphasis> mode. This option only
works when bridging a DAHDI channel to another DAHDI channel
only. if specified on non-DAHDI interfaces, it will be ignored.
When the destination answers (presumably an operator services
station), the originator no longer has control of their line.
They may hang up, but the switch will not release their line
until the destination party (the operator) hangs up.</para>
</option>
<option name="p">
<para>This option enables screening mode. This is basically Privacy mode
without memory.</para>
</option>
<option name="P">
<argument name="x" />
<para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
it is provided. The current extension is used if a database family/key is not specified.</para>
</option>
<option name="Q">
<argument name="cause" required="true"/>
<para>Specify the Q.850/Q.931 <replaceable>cause</replaceable> to send on
unanswered channels when another channel answers the call.
As with <literal>Hangup()</literal>, <replaceable>cause</replaceable>
can be a numeric cause code or a name such as
<literal>NO_ANSWER</literal>,
<literal>USER_BUSY</literal>,
<literal>CALL_REJECTED</literal> or
<literal>ANSWERED_ELSEWHERE</literal> (the default if Q isn't specified).
You can also specify <literal>0</literal> or <literal>NONE</literal>
to send no cause. See the <filename>causes.h</filename> file for the
full list of valid causes and names.
</para>
<note>
<para>chan_sip does not support setting the cause on a CANCEL to anything
other than ANSWERED_ELSEWHERE.</para>
</note>
</option>
<option name="r">
<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
party until the called channel has answered.</para>
<argument name="tone" required="false">
<para>Indicate progress to calling party. Send audio 'tone' from the <filename>indications.conf</filename> tonezone currently in use.</para>
</argument>
</option>
<option name="R">
<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
Allow interruption of the ringback if early media is received on the channel.</para>
</option>
<option name="S">
<argument name="x" required="true" />
<para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
answered the call.</para>
</option>
<option name="s">
<argument name="x" required="true" />
<para>Force the outgoing CallerID tag parameter to be set to the string <replaceable>x</replaceable>.</para>
<para>Works with the <literal>f</literal> option.</para>
</option>
<option name="t">
<para>Allow the called party to transfer the calling party by sending the
DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
transfers initiated by other methods.</para>
</option>
<option name="T">
<para>Allow the calling party to transfer the called party by sending the
DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
transfers initiated by other methods.</para>
</option>
<option name="U" argsep="^">
<argument name="x" required="true">
<para>Name of the subroutine to execute via <literal>Gosub</literal></para>
</argument>
<argument name="arg" multiple="true" required="false">
<para>Arguments for the <literal>Gosub</literal> routine</para>
</argument>
<para>Execute via <literal>Gosub</literal> the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
to the calling channel. Arguments can be specified to the <literal>Gosub</literal>
using <literal>^</literal> as a delimiter. The <literal>Gosub</literal> routine can set the variable
<variable>GOSUB_RESULT</variable> to specify the following actions after the <literal>Gosub</literal> returns.</para>
<variablelist>
<variable name="GOSUB_RESULT">
<value name="ABORT">
Hangup both legs of the call.
</value>
<value name="CONGESTION">
Behave as if line congestion was encountered.
</value>
<value name="BUSY">
Behave as if a busy signal was encountered.
</value>
<value name="CONTINUE">
Hangup the called party and allow the calling party
to continue dialplan execution at the next priority.
</value>
<value name="GOTO:[[&lt;context&gt;^]&lt;exten&gt;^]&lt;priority&gt;">
Transfer the call to the specified destination.
</value>
</variable>
</variablelist>
<note>
<para>You cannot use any additional action post answer options in conjunction
with this option. Also, pbx services are run on the peer (called) channel,
so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
</note>
</option>
<option name="u">
<argument name = "x" required="true">
<para>Force the outgoing callerid presentation indicator parameter to be set
to one of the values passed in <replaceable>x</replaceable>:
<literal>allowed_not_screened</literal>
<literal>allowed_passed_screen</literal>
<literal>allowed_failed_screen</literal>
<literal>allowed</literal>
<literal>prohib_not_screened</literal>
<literal>prohib_passed_screen</literal>
<literal>prohib_failed_screen</literal>
<literal>prohib</literal>
<literal>unavailable</literal>
</para>
</argument>
<para>Works with the <literal>f</literal> option.</para>
</option>
<option name="w">
<para>Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="W">
<para>Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="x">
<para>Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
</option>
<option name="X">
<para>Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
</option>
<option name="z">
<para>On a call forward, cancel any dial timeout which has been set for this call.</para>
</option>
</optionlist>
</parameter>
<parameter name="URL">
<para>The optional URL will be sent to the called party if the channel driver supports it.</para>
</parameter>
</syntax>
<description>
<para>This application will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
be active in a bridged call. All other channels that were requested will then
be hung up.</para>
<para>Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan execution will
continue if no requested channels can be called, or if the timeout expires.
This application will report normal termination if the originating channel
hangs up, or if the call is bridged and either of the parties in the bridge
ends the call.</para>
<para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
application will be put into that group (as in <literal>Set(GROUP()=...</literal>).
If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
however, the variable will be unset after use.</para>
<example title="Dial with 30 second timeout">
same => n,Dial(PJSIP/alice,30)
</example>
<example title="Parallel dial with 45 second timeout">
same => n,Dial(PJSIP/alice&amp;PJIP/bob,45)
</example>
<example title="Dial with 'g' continuation option">
same => n,Dial(PJSIP/alice,,g)
same => n,Log(NOTICE, Alice call result: ${DIALSTATUS})
</example>
<example title="Dial with transfer/recording features for calling party">
same => n,Dial(PJSIP/alice,,TX)
</example>
<example title="Dial with call length limit">
same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
</example>
<example title="Dial alice and bob and send NO_ANSWER to bob instead of ANSWERED_ELSEWHERE when alice answers">
same => n,Dial(PJSIP/alice&amp;PJSIP/bob,,Q(NO_ANSWER))
</example>
<example title="Dial with pre-dial subroutines">
[default]
exten => callee_channel,1,NoOp()
same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
same => n,Return()
exten => called_channel,1,NoOp()
same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
same => n,Return()
exten => _X.,1,NoOp()
same => n,Dial(PJSIP/alice,,b(default^called_channel^1)B(default^callee_channel^1))
same => n,Hangup()
</example>
<example title="Dial with post-answer subroutine executed on outbound channel">
[default]
exten => called_channel,1,NoOp()
same => n,Playback(hello)
same => n,Return()
exten => _X.,1,NoOp()
same => n,Dial(PJSIP/alice,,U(default^called_channel^1))
same => n,Hangup()
</example>
<example title="Dial into ConfBridge using 'G' option">
same => n,Dial(PJSIP/alice,,G(jump_to_here))
same => n(jump_to_here),Goto(confbridge)
same => n,Goto(confbridge)
same => n(confbridge),ConfBridge(${EXTEN})
</example>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="DIALEDTIME">
<para>This is the time from dialing a channel until when it is disconnected.</para>
</variable>
<variable name="ANSWEREDTIME">
<para>This is the amount of time for actual call.</para>
</variable>
<variable name="DIALEDPEERNAME">
<para>The name of the outbound channel that answered the call.</para>
</variable>
<variable name="DIALEDPEERNUMBER">
<para>The number that was dialed for the answered outbound channel.</para>
</variable>
<variable name="FORWARDERNAME">
<para>If a call forward occurred, the name of the forwarded channel.</para>
</variable>
<variable name="DIALSTATUS">
<para>This is the status of the call</para>
<value name="CHANUNAVAIL" />
<value name="CONGESTION" />
<value name="NOANSWER" />
<value name="BUSY" />
<value name="ANSWER" />
<value name="CANCEL" />
<value name="DONTCALL">
For the Privacy and Screening Modes.
Will be set if the called party chooses to send the calling party to the 'Go Away' script.
</value>
<value name="TORTURE">
For the Privacy and Screening Modes.
Will be set if the called party chooses to send the calling party to the 'torture' script.
</value>
<value name="INVALIDARGS" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">RetryDial</ref>
<ref type="application">SendDTMF</ref>
<ref type="application">Gosub</ref>
<ref type="application">Macro</ref>
</see-also>
</application>
<application name="RetryDial" language="en_US">
<synopsis>
Place a call, retrying on failure allowing an optional exit extension.
</synopsis>
<syntax>
<parameter name="announce" required="true">
<para>Filename of sound that will be played when no channel can be reached</para>
</parameter>
<parameter name="sleep" required="true">
<para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
</parameter>
<parameter name="retries" required="true">
<para>Number of retries</para>
<para>When this is reached flow will continue at the next priority in the dialplan</para>
</parameter>
<parameter name="dialargs" required="true">
<para>Same format as arguments provided to the Dial application</para>
</parameter>
</syntax>
<description>
<para>This application will attempt to place a call using the normal Dial application.
If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
While waiting to retry a call, a 1 digit extension may be dialed. If that
extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
one, The call will jump to that extension immediately.
The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
to the Dial application.</para>
</description>
<see-also>
<ref type="application">Dial</ref>
</see-also>
</application>
<application name="SkelGuessNumber" language="en_US">
<synopsis>
An example number guessing game
</synopsis>
<syntax>
<parameter name="level" required="true"/>
<parameter name="options">
<optionlist>
<option name="c">
<para>The computer should cheat</para>
</option>
<option name="n">
<para>How many games to play before hanging up</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This simple number guessing application is a template to build other applications
from. It shows you the basic structure to create your own Asterisk applications.</para>
</description>
</application>
<configInfo name="app_skel" language="en_US">
<configFile name="app_skel.conf">
<configObject name="globals">
<synopsis>Options that apply globally to app_skel</synopsis>
<configOption name="games">
<synopsis>The number of games a single execution of SkelGuessNumber will play</synopsis>
</configOption>
<configOption name="cheat">
<synopsis>Should the computer cheat?</synopsis>
<description>
<para>If enabled, the computer will ignore winning guesses.</para>
</description>
</configOption>
</configObject>
<configObject name="sounds">
<synopsis>Prompts for SkelGuessNumber to play</synopsis>
<configOption name="prompt" default="please-enter-your&amp;number&amp;queue-less-than">
<synopsis>A prompt directing the user to enter a number less than the max number</synopsis>
</configOption>
<configOption name="wrong_guess" default="vm-pls-try-again">
<synopsis>The sound file to play when a wrong guess is made</synopsis>
</configOption>
<configOption name="right_guess" default="auth-thankyou">
<synopsis>The sound file to play when a correct guess is made</synopsis>
</configOption>
<configOption name="too_low">
<synopsis>The sound file to play when a guess is too low</synopsis>
</configOption>
<configOption name="too_high">
<synopsis>The sound file to play when a guess is too high</synopsis>
</configOption>
<configOption name="lose" default="vm-goodbye">
<synopsis>The sound file to play when a player loses</synopsis>
</configOption>
</configObject>
<configObject name="level">
<synopsis>Defined levels for the SkelGuessNumber game</synopsis>
<configOption name="max_number">
<synopsis>The maximum in the range of numbers to guess (1 is the implied minimum)</synopsis>
</configOption>
<configOption name="max_guesses">
<synopsis>The maximum number of guesses before a game is considered lost</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<application name="SoftHangup" language="en_US">
<synopsis>
Hangs up the requested channel.
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" />
<parameter name="options">
<optionlist>
<option name="a">
<para>Hang up all channels on a specified device instead of a single resource</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Hangs up the requested channel. If there are no channels to
hangup, the application will report it.</para>
</description>
</application>
<application name="NoCDR" language="en_US">
<synopsis>
Tell Asterisk to not maintain a CDR for this channel.
</synopsis>
<syntax />
<description>
<para>This application will tell Asterisk not to maintain a CDR for
the current channel. This does <emphasis>NOT</emphasis> mean that
information is not tracked; rather, if the channel is hung up no
CDRs will be created for that channel.</para>
<para>If a subsequent call to ResetCDR occurs, all non-finalized
CDRs created for the channel will be enabled.</para>
<note>
<para>This application is deprecated. Please use the CDR_PROP
function to disable CDRs on a channel.</para>
</note>
</description>
<see-also>
<ref type="application">ResetCDR</ref>
<ref type="function">CDR_PROP</ref>
</see-also>
</application>
<application name="ResetCDR" language="en_US">
<synopsis>
Resets the Call Data Record.
</synopsis>
<syntax>
<parameter name="options">
<optionlist>
<option name="v">
<para>Save the CDR variables during the reset.</para>
</option>
<option name="e">
<para>Enable the CDRs for this channel only (negate
effects of NoCDR).</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application causes the Call Data Record to be reset.
Depending on the flags passed in, this can have several effects.
With no options, a reset does the following:</para>
<para>1. The <literal>start</literal> time is set to the current time.</para>
<para>2. If the channel is answered, the <literal>answer</literal> time is set to the
current time.</para>
<para>3. All variables are wiped from the CDR. Note that this step
can be prevented with the <literal>v</literal> option.</para>
<para>On the other hand, if the <literal>e</literal> option is
specified, the effects of the NoCDR application will be lifted. CDRs
will be re-enabled for this channel.</para>
<note>
<para>The <literal>e</literal> option is deprecated. Please
use the CDR_PROP function instead.</para>
</note>
</description>
<see-also>
<ref type="application">ForkCDR</ref>
<ref type="application">NoCDR</ref>
<ref type="function">CDR_PROP</ref>
</see-also>
</application>
<application name="SendText" language="en_US">
<synopsis>
Send a Text Message.
</synopsis>
<syntax>
<parameter name="text" required="true" />
</syntax>
<description>
<para>Sends <replaceable>text</replaceable> to current channel (callee).</para>
<para>Result of transmission will be stored in the <variable>SENDTEXTSTATUS</variable></para>
<variablelist>
<variable name="SENDTEXTSTATUS">
<value name="SUCCESS">
Transmission succeeded.
</value>
<value name="FAILURE">
Transmission failed.
</value>
<value name="UNSUPPORTED">
Text transmission not supported by channel.
</value>
</variable>
</variablelist>
<note>
<para>At this moment, text is supposed to be 7 bit ASCII in most channels.</para>
</note>
</description>
<see-also>
<ref type="application">SendImage</ref>
<ref type="application">SendURL</ref>
</see-also>
</application>
<application name="IVRDemo" language="en_US">
<synopsis>
IVR Demo Application.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
</syntax>
<description>
<para>This is a skeleton application that shows you the basic structure to create your
own asterisk applications and demonstrates the IVR demo.</para>
</description>
</application>
<application name="AgentLogin" language="en_US">
<synopsis>
Login an agent.
</synopsis>
<syntax argsep=",">
<parameter name="AgentId" required="true" />
<parameter name="options">
<optionlist>
<option name="s">
<para>silent login - do not announce the login ok segment after
agent logged on.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Login an agent to the system. Any agent authentication is assumed to
already be done by dialplan. While logged in, the agent can receive calls
and will hear the sound file specified by the config option custom_beep
when a new call comes in for the agent. Login failures will continue in
the dialplan with <variable>AGENT_STATUS</variable> set.</para>
<para>Before logging in, you can setup on the real agent channel the
CHANNEL(dtmf-features) an agent will have when talking to a caller
and you can setup on the channel running this application the
CONNECTEDLINE() information the agent will see while waiting for a
caller.</para>
<para>
<variable>AGENT_STATUS</variable> enumeration values:</para>
<enumlist>
<enum name = "INVALID">
<para>The specified agent is invalid.</para>
</enum>
<enum name = "ALREADY_LOGGED_IN">
<para>The agent is already logged in.</para>
</enum>
</enumlist>
<note>
<para>The Agents:<replaceable>AgentId</replaceable> device state is
available to monitor the status of the agent.</para>
</note>
</description>
<see-also>
<ref type="application">Authenticate</ref>
<ref type="application">Queue</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">AGENT</ref>
<ref type="function">CHANNEL(dtmf-features)</ref>
<ref type="function">CONNECTEDLINE()</ref>
<ref type="filename">agents.conf</ref>
<ref type="filename">queues.conf</ref>
</see-also>
</application>
<application name="AgentRequest" language="en_US">
<synopsis>
Request an agent to connect with the channel.
</synopsis>
<syntax argsep=",">
<parameter name="AgentId" required="true" />
</syntax>
<description>
<para>Request an agent to connect with the channel. Failure to find,
alert the agent, or acknowledge the call will continue in the dialplan
with <variable>AGENT_STATUS</variable> set.</para>
<para>
<variable>AGENT_STATUS</variable> enumeration values:</para>
<enumlist>
<enum name = "INVALID">
<para>The specified agent is invalid.</para>
</enum>
<enum name = "NOT_LOGGED_IN">
<para>The agent is not available.</para>
</enum>
<enum name = "BUSY">
<para>The agent is on another call.</para>
</enum>
<enum name = "NOT_CONNECTED">
<para>The agent did not connect with the
call. The agent most likely did not acknowledge the call.</para>
</enum>
<enum name = "ERROR">
<para>Alerting the agent failed.</para>
</enum>
</enumlist>
</description>
<see-also>
<ref type="application">AgentLogin</ref>
</see-also>
</application>
<function name="AGENT" language="en_US">
<synopsis>
Gets information about an Agent
</synopsis>
<syntax argsep=":">
<parameter name="AgentId" required="true" />
<parameter name="item">
<para>The valid items to retrieve are:</para>
<enumlist>
<enum name="status">
<para>(default) The status of the agent (LOGGEDIN | LOGGEDOUT)</para>
</enum>
<enum name="password">
<para>Deprecated. The dialplan handles any agent authentication.</para>
</enum>
<enum name="name">
<para>The name of the agent</para>
</enum>
<enum name="mohclass">
<para>MusicOnHold class</para>
</enum>
<enum name="channel">
<para>The name of the active channel for the Agent (AgentLogin)</para>
</enum>
<enum name="fullchannel">
<para>The untruncated name of the active channel for the Agent (AgentLogin)</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description></description>
</function>
<manager name="Agents" language="en_US">
<synopsis>
Lists agents and their status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Will list info about all defined agents.</para>
</description>
<see-also>
<ref type="managerEvent">Agents</ref>
<ref type="managerEvent">AgentsComplete</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="Agents">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>
Response event in a series to the Agents AMI action containing
information about a defined agent.
</synopsis>
<syntax>
<parameter name="Agent">
<para>Agent ID of the agent.</para>
</parameter>
<parameter name="Name">
<para>User friendly name of the agent.</para>
</parameter>
<parameter name="Status">
<para>Current status of the agent.</para>
<para>The valid values are:</para>
<enumlist>
<enum name="AGENT_LOGGEDOFF" />
<enum name="AGENT_IDLE" />
<enum name="AGENT_ONCALL" />
</enumlist>
</parameter>
<parameter name="TalkingToChan">
<para>
<variable>BRIDGEPEER</variable> value on agent channel.</para>
<para>Present if Status value is <literal>AGENT_ONCALL</literal>.</para>
</parameter>
<parameter name="CallStarted">
<para>Epoche time when the agent started talking with the caller.</para>
<para>Present if Status value is <literal>AGENT_ONCALL</literal>.</para>
</parameter>
<parameter name="LoggedInTime">
<para>Epoche time when the agent logged in.</para>
<para>Present if Status value is <literal>AGENT_IDLE</literal> or <literal>AGENT_ONCALL</literal>.</para>
</parameter>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>The channel snapshot is present if the Status value is <literal>AGENT_IDLE</literal> or <literal>AGENT_ONCALL</literal>.</para>
</description>
<see-also>
<ref type="manager">Agents</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentsComplete">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>
Final response event in a series of events to the Agents AMI action.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<see-also>
<ref type="manager">Agents</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="AgentLogoff" language="en_US">
<synopsis>
Sets an agent as no longer logged in.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Agent" required="true">
<para>Agent ID of the agent to log off.</para>
</parameter>
<parameter name="Soft">
<para>Set to <literal>true</literal> to not hangup existing calls.</para>
</parameter>
</syntax>
<description>
<para>Sets an agent as no longer logged in.</para>
</description>
</manager>
<configInfo name="app_agent_pool" language="en_US">
<synopsis>Agent pool applications</synopsis>
<description>
<note>
<para>Option changes take effect on agent login or after an agent
disconnects from a call.</para>
</note>
</description>
<configFile name="agents.conf">
<configObject name="global">
<synopsis>Unused, but reserved.</synopsis>
</configObject>
<configObject name="agent-id">
<synopsis>Configure an agent for the pool.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
<configOption name="ackcall">
<synopsis>Enable to require the agent to acknowledge a call.</synopsis>
<description>
<para>Enable to require the agent to give a DTMF acknowledgement
when the agent receives a call.</para>
<note>
<para>The option is overridden by <variable>AGENTACKCALL</variable> on agent login.</para>
</note>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="acceptdtmf">
<synopsis>DTMF key sequence the agent uses to acknowledge a call.</synopsis>
<description>
<note>
<para>The option is overridden by <variable>AGENTACCEPTDTMF</variable> on agent login.</para>
</note>
<note>
<para>The option is ignored unless the ackcall option is enabled.</para>
</note>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="autologoff">
<synopsis>Time the agent has to acknowledge a call before being logged off.</synopsis>
<description>
<para>Set how many seconds a call for the agent has to wait for the
agent to acknowledge the call before the agent is automatically
logged off. If set to zero then the call will wait forever for
the agent to acknowledge.</para>
<note>
<para>The option is overridden by <variable>AGENTAUTOLOGOFF</variable> on agent login.</para>
</note>
<note>
<para>The option is ignored unless the ackcall option is enabled.</para>
</note>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="wrapuptime">
<synopsis>Minimum time the agent has between calls.</synopsis>
<description>
<para>Set the minimum amount of time in milliseconds after
disconnecting a call before the agent can receive a new call.</para>
<note>
<para>The option is overridden by <variable>AGENTWRAPUPTIME</variable> on agent login.</para>
</note>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="musiconhold">
<synopsis>Music on hold class the agent listens to between calls.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="recordagentcalls">
<synopsis>Enable to automatically record calls the agent takes.</synopsis>
<description>
<para>Enable recording calls the agent takes automatically by
invoking the automixmon DTMF feature when the agent connects
to a caller. See <filename>features.conf.sample</filename> for information about
the automixmon feature.</para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="custom_beep">
<synopsis>Sound file played to alert the agent when a call is present.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
<configOption name="fullname">
<synopsis>A friendly name for the agent used in log messages.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='app_agent_pool']/description/note)" />
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<application name="MP3Player" language="en_US">
<synopsis>
Play an MP3 file or M3U playlist file or stream.
</synopsis>
<syntax>
<parameter name="Location" required="true">
<para>Location of the file to be played.
(argument passed to mpg123)</para>
</parameter>
</syntax>
<description>
<para>Executes mpg123 to play the given location, which typically would be a mp3 filename
or m3u playlist filename or a URL. Please read http://en.wikipedia.org/wiki/M3U
to see how M3U playlist file format is like, Example usage would be
exten => 1234,1,MP3Player(/var/lib/asterisk/playlist.m3u)
User can exit by pressing any key on the dialpad, or by hanging up.</para>
<para>This application does not automatically answer and should be preceeded by an
application such as Answer() or Progress().</para>
</description>
</application>
<application name="VoiceMail" language="en_US">
<synopsis>
Leave a Voicemail message.
</synopsis>
<syntax>
<parameter name="mailboxs" argsep="&amp;" required="true">
<argument name="mailbox1" argsep="@" required="true">
<argument name="mailbox" required="true" />
<argument name="context" />
</argument>
<argument name="mailbox2" argsep="@" multiple="true">
<argument name="mailbox" required="true" />
<argument name="context" />
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="b">
<para>Play the <literal>busy</literal> greeting to the calling party.</para>
</option>
<option name="d">
<argument name="c" />
<para>Accept digits for a new extension in context <replaceable>c</replaceable>,
if played during the greeting. Context defaults to the current context.</para>
</option>
<option name="g">
<argument name="#" required="true" />
<para>Use the specified amount of gain when recording the voicemail
message. The units are whole-number decibels (dB). Only works on supported
technologies, which is DAHDI only.</para>
</option>
<option name="s">
<para>Skip the playback of instructions for leaving a message to the
calling party.</para>
</option>
<option name="u">
<para>Play the <literal>unavailable</literal> greeting.</para>
</option>
<option name="U">
<para>Mark message as <literal>URGENT</literal>.</para>
</option>
<option name="P">
<para>Mark message as <literal>PRIORITY</literal>.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application allows the calling party to leave a message for the specified
list of mailboxes. When multiple mailboxes are specified, the greeting will be taken from
the first mailbox specified. Dialplan execution will stop if the specified mailbox does not
exist.</para>
<para>The Voicemail application will exit if any of the following DTMF digits are received:</para>
<enumlist>
<enum name="0">
<para>Jump to the <literal>o</literal> extension in the current dialplan context.</para>
</enum>
<enum name="*">
<para>Jump to the <literal>a</literal> extension in the current dialplan context.</para>
</enum>
</enumlist>
<para>This application will set the following channel variable upon completion:</para>
<variablelist>
<variable name="VMSTATUS">
<para>This indicates the status of the execution of the VoiceMail application.</para>
<value name="SUCCESS" />
<value name="USEREXIT" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">VoiceMailMain</ref>
</see-also>
</application>
<application name="VoiceMailMain" language="en_US">
<synopsis>
Check Voicemail messages.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="mailbox" />
<argument name="context" />
</parameter>
<parameter name="options">
<optionlist>
<option name="p">
<para>Consider the <replaceable>mailbox</replaceable> parameter as a prefix to
the mailbox that is entered by the caller.</para>
</option>
<option name="g">
<argument name="#" required="true" />
<para>Use the specified amount of gain when recording a voicemail message.
The units are whole-number decibels (dB).</para>
</option>
<option name="s">
<para>Skip checking the passcode for the mailbox.</para>
</option>
<option name="a">
<argument name="folder" required="true" />
<para>Skip folder prompt and go directly to <replaceable>folder</replaceable> specified.
Defaults to <literal>INBOX</literal> (or <literal>0</literal>).</para>
<enumlist>
<enum name="0">
<para>INBOX</para>
</enum>
<enum name="1">
<para>Old</para>
</enum>
<enum name="2">
<para>Work</para>
</enum>
<enum name="3">
<para>Family</para>
</enum>
<enum name="4">
<para>Friends</para>
</enum>
<enum name="5">
<para>Cust1</para>
</enum>
<enum name="6">
<para>Cust2</para>
</enum>
<enum name="7">
<para>Cust3</para>
</enum>
<enum name="8">
<para>Cust4</para>
</enum>
<enum name="9">
<para>Cust5</para>
</enum>
</enumlist>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application allows the calling party to check voicemail messages. A specific
<replaceable>mailbox</replaceable>, and optional corresponding <replaceable>context</replaceable>,
may be specified. If a <replaceable>mailbox</replaceable> is not provided, the calling party will
be prompted to enter one. If a <replaceable>context</replaceable> is not specified, the
<literal>default</literal> context will be used.</para>
<para>The VoiceMailMain application will exit if the following DTMF digit is entered as Mailbox
or Password, and the extension exists:</para>
<enumlist>
<enum name="*">
<para>Jump to the <literal>a</literal> extension in the current dialplan context.</para>
</enum>
</enumlist>
</description>
<see-also>
<ref type="application">VoiceMail</ref>
</see-also>
</application>
<application name="MailboxExists" language="en_US">
<synopsis>
Check to see if Voicemail mailbox exists.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="mailbox" required="true" />
<argument name="context" />
</parameter>
<parameter name="options">
<para>None options.</para>
</parameter>
</syntax>
<description>
<note>
<para>DEPRECATED. Use VM_INFO(mailbox[@context],exists) instead.</para>
</note>
<para>Check to see if the specified <replaceable>mailbox</replaceable> exists. If no voicemail
<replaceable>context</replaceable> is specified, the <literal>default</literal> context
will be used.</para>
<para>This application will set the following channel variable upon completion:</para>
<variablelist>
<variable name="VMBOXEXISTSSTATUS">
<para>This will contain the status of the execution of the MailboxExists application.
Possible values include:</para>
<value name="SUCCESS" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="function">VM_INFO</ref>
</see-also>
</application>
<application name="VMAuthenticate" language="en_US">
<synopsis>
Authenticate with Voicemail passwords.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="mailbox" />
<argument name="context" />
</parameter>
<parameter name="options">
<optionlist>
<option name="s">
<para>Skip playing the initial prompts.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application behaves the same way as the Authenticate application, but the passwords
are taken from <filename>voicemail.conf</filename>. If the <replaceable>mailbox</replaceable> is
specified, only that mailbox's password will be considered valid. If the <replaceable>mailbox</replaceable>
is not specified, the channel variable <variable>AUTH_MAILBOX</variable> will be set with the authenticated
mailbox.</para>
<para>The VMAuthenticate application will exit if the following DTMF digit is entered as Mailbox
or Password, and the extension exists:</para>
<enumlist>
<enum name="*">
<para>Jump to the <literal>a</literal> extension in the current dialplan context.</para>
</enum>
</enumlist>
</description>
</application>
<application name="VoiceMailPlayMsg" language="en_US">
<synopsis>
Play a single voice mail msg from a mailbox by msg id.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="mailbox" />
<argument name="context" />
</parameter>
<parameter name="msg_id" required="true">
<para>The msg id of the msg to play back. </para>
</parameter>
</syntax>
<description>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="VOICEMAIL_PLAYBACKSTATUS">
<para>The status of the playback attempt as a text string.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
</description>
</application>
<application name="VMSayName" language="en_US">
<synopsis>
Play the name of a voicemail user
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="mailbox" />
<argument name="context" />
</parameter>
</syntax>
<description>
<para>This application will say the recorded name of the voicemail user specified as the
argument to this application. If no context is provided, <literal>default</literal> is assumed.</para>
</description>
</application>
<function name="MAILBOX_EXISTS" language="en_US">
<synopsis>
Tell if a mailbox is configured.
</synopsis>
<syntax argsep="@">
<parameter name="mailbox" required="true" />
<parameter name="context" />
</syntax>
<description>
<note>
<para>DEPRECATED. Use VM_INFO(mailbox[@context],exists) instead.</para>
</note>
<para>Returns a boolean of whether the corresponding <replaceable>mailbox</replaceable> exists.
If <replaceable>context</replaceable> is not specified, defaults to the <literal>default</literal>
context.</para>
</description>
<see-also>
<ref type="function">VM_INFO</ref>
</see-also>
</function>
<function name="VM_INFO" language="en_US">
<synopsis>
Returns the selected attribute from a mailbox.
</synopsis>
<syntax argsep=",">
<parameter name="mailbox" argsep="@" required="true">
<argument name="mailbox" required="true" />
<argument name="context" />
</parameter>
<parameter name="attribute" required="true">
<optionlist>
<option name="count">
<para>Count of messages in specified <replaceable>folder</replaceable>.
If <replaceable>folder</replaceable> is not specified, defaults to <literal>INBOX</literal>.</para>
</option>
<option name="email">
<para>E-mail address associated with the mailbox.</para>
</option>
<option name="exists">
<para>Returns a boolean of whether the corresponding <replaceable>mailbox</replaceable> exists.</para>
</option>
<option name="fullname">
<para>Full name associated with the mailbox.</para>
</option>
<option name="language">
<para>Mailbox language if overridden, otherwise the language of the channel.</para>
</option>
<option name="locale">
<para>Mailbox locale if overridden, otherwise global locale.</para>
</option>
<option name="pager">
<para>Pager e-mail address associated with the mailbox.</para>
</option>
<option name="password">
<para>Mailbox access password.</para>
</option>
<option name="tz">
<para>Mailbox timezone if overridden, otherwise global timezone</para>
</option>
</optionlist>
</parameter>
<parameter name="folder" required="false">
<para>If not specified, <literal>INBOX</literal> is assumed.</para>
</parameter>
</syntax>
<description>
<para>Returns the selected attribute from the specified <replaceable>mailbox</replaceable>.
If <replaceable>context</replaceable> is not specified, defaults to the <literal>default</literal>
context. Where the <replaceable>folder</replaceable> can be specified, common folders
include <literal>INBOX</literal>, <literal>Old</literal>, <literal>Work</literal>,
<literal>Family</literal> and <literal>Friends</literal>.</para>
</description>
</function>
<manager name="VoicemailUsersList" language="en_US">
<synopsis>
List All Voicemail User Information.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
</description>
</manager>
<manager name="VoicemailRefresh" language="en_US">
<synopsis>
Tell Asterisk to poll mailboxes for a change
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Context" />
<parameter name="Mailbox" />
</syntax>
<description>
<para>Normally, MWI indicators are only sent when Asterisk itself
changes a mailbox. With external programs that modify the content
of a mailbox from outside the application, an option exists called
<literal>pollmailboxes</literal> that will cause voicemail to
continually scan all mailboxes on a system for changes. This can
cause a large amount of load on a system. This command allows
external applications to signal when a particular mailbox has
changed, thus permitting external applications to modify mailboxes
and MWI to work without introducing considerable CPU load.</para>
<para>If <replaceable>Context</replaceable> is not specified, all
mailboxes on the system will be polled for changes. If
<replaceable>Context</replaceable> is specified, but
<replaceable>Mailbox</replaceable> is omitted, then all mailboxes
within <replaceable>Context</replaceable> will be polled.
Otherwise, only a single mailbox will be polled for changes.</para>
</description>
</manager>
<application name="ForkCDR" language="en_US">
<synopsis>
Forks the current Call Data Record for this channel.
</synopsis>
<syntax>
<parameter name="options">
<optionlist>
<option name="a">
<para>If the channel is answered, set the answer time on
the forked CDR to the current time. If this option is
not used, the answer time on the forked CDR will be the
answer time on the original CDR. If the channel is not
answered, this option has no effect.</para>
<para>Note that this option is implicitly assumed if the
<literal>r</literal> option is used.</para>
</option>
<option name="e">
<para>End (finalize) the original CDR.</para>
</option>
<option name="r">
<para>Reset the start and answer times on the forked CDR.
This will set the start and answer times (if the channel
is answered) to be set to the current time.</para>
<para>Note that this option implicitly assumes the
<literal>a</literal> option.</para>
</option>
<option name="v">
<para>Do not copy CDR variables and attributes from the
original CDR to the forked CDR.</para>
<warning>
<para>This option has changed. Previously, the
variables were removed from the original CDR. This no
longer occurs - this option now controls whether or not
a forked CDR inherits the variables from the original
CDR.</para>
</warning>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Causes the Call Data Record engine to fork a new CDR starting
from the time the application is executed. The forked CDR will be
linked to the end of the CDRs associated with the channel.</para>
</description>
<see-also>
<ref type="function">CDR</ref>
<ref type="application">NoCDR</ref>
<ref type="application">ResetCDR</ref>
</see-also>
</application>
<application name="Echo" language="en_US">
<synopsis>
Echo media, DTMF back to the calling party
</synopsis>
<syntax />
<description>
<para>Echos back any media or DTMF frames read from the calling
channel back to itself. This will not echo CONTROL, MODEM, or NULL
frames. Note: If '#' detected application exits.</para>
<para>This application does not automatically answer and should be
preceeded by an application such as Answer() or Progress().</para>
</description>
</application>
<application name="SayCountedNoun" language="en_US">
<synopsis>
Say a noun in declined form in order to count things
</synopsis>
<syntax>
<parameter name="number" required="true">
<para>The number of things</para>
</parameter>
<parameter name="filename" required="true">
<para>File name stem for the noun that is the the name of the things</para>
</parameter>
</syntax>
<description>
<para>Selects and plays the proper singular or plural form of a noun
when saying things such as "five calls". English has simple rules
for deciding when to say "call" and when to say "calls", but other
languages have complicated rules which would be extremely difficult
to implement in the Asterisk dialplan language.</para>
<para>The correct sound file is selected by examining the
<replaceable>number</replaceable> and adding the appropriate suffix
to <replaceable>filename</replaceable>. If the channel language is
English, then the suffix will be either empty or "s". If the channel
language is Russian or some other Slavic language, then the suffix
will be empty for nominative, "x1" for genative singular, and "x2"
for genative plural.</para>
<para>Note that combining <replaceable>filename</replaceable> with
a suffix will not necessarily produce a correctly spelled plural
form. For example, SayCountedNoun(2,man) will play the sound file
"mans" rather than "men". This behavior is intentional. Since the
file name is never seen by the end user, there is no need to
implement complicated spelling rules. We simply record the word
"men" in the sound file named "mans".</para>
<para>This application does not automatically answer and should be
preceeded by an application such as Answer() or Progress.</para>
</description>
<see-also>
<ref type="application">SayCountedAdj</ref>
<ref type="application">SayNumber</ref>
</see-also>
</application>
<application name="SayCountedAdj" language="en_US">
<synopsis>
Say a adjective in declined form in order to count things
</synopsis>
<syntax>
<parameter name="number" required="true">
<para>The number of things</para>
</parameter>
<parameter name="filename" required="true">
<para>File name stem for the adjective</para>
</parameter>
<parameter name="gender">
<para>The gender of the noun modified, one of 'm', 'f', 'n', or 'c'</para>
</parameter>
</syntax>
<description>
<para>Selects and plays the proper form of an adjective according to
the gender and of the noun which it modifies and the number of
objects named by the noun-verb combination which have been counted.
Used when saying things such as "5 new messages". The various
singular and plural forms of the adjective are selected by adding
suffixes to <replaceable>filename</replaceable>.</para>
<para>If the channel language is English, then no suffix will ever
be added (since, in English, adjectives are not declined). If the
channel language is Russian or some other slavic language, then the
suffix will the specified <replaceable>gender</replaceable> for
nominative, and "x" for genative plural. (The genative singular is
not used when counting things.) For example, SayCountedAdj(1,new,f)
will play sound file "newa" (containing the word "novaya"), but
SayCountedAdj(5,new,f) will play sound file "newx" (containing the
word "novikh").</para>
<para>This application does not automatically answer and should be
preceeded by an application such as Answer(), Progress(), or
Proceeding().</para>
</description>
<see-also>
<ref type="application">SayCountedNoun</ref>
<ref type="application">SayNumber</ref>
</see-also>
</application>
<application name="ChanIsAvail" language="en_US">
<synopsis>
Check channel availability
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to check</para>
<para>If you need more than one enter them as
Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
</argument>
<para>Specification of the device(s) to check. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable>
represents a resource available to that particular channel driver.</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="a">
<para>Check for all available channels, not only the first one</para>
</option>
<option name="s">
<para>Consider the channel unavailable if the channel is in use at all</para>
</option>
<option name="t" implies="s">
<para>Simply checks if specified channels exist in the channel list</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application will check to see if any of the specified channels are available.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="AVAILCHAN">
<para>The name of the available channel, if one exists</para>
</variable>
<variable name="AVAILORIGCHAN">
<para>The canonical channel name that was used to create the channel</para>
</variable>
<variable name="AVAILSTATUS">
<para>The device state for the device</para>
</variable>
<variable name="AVAILCAUSECODE">
<para>The cause code returned when requesting the channel</para>
</variable>
</variablelist>
</description>
</application>
<application name="DumpChan" language="en_US">
<synopsis>
Dump Info About The Calling Channel.
</synopsis>
<syntax>
<parameter name="level">
<para>Minimum verbose level</para>
</parameter>
</syntax>
<description>
<para>Displays information on channel and listing of all channel
variables. If <replaceable>level</replaceable> is specified, output is only
displayed when the verbose level is currently set to that number
or greater.</para>
</description>
<see-also>
<ref type="application">NoOp</ref>
<ref type="application">Verbose</ref>
</see-also>
</application>
<application name="GetCPEID" language="en_US">
<synopsis>
Get ADSI CPE ID.
</synopsis>
<syntax />
<description>
<para>Obtains and displays ADSI CPE ID and other information in order
to properly setup <filename>dahdi.conf</filename> for on-hook operations.</para>
</description>
</application>
<application name="Record" language="en_US">
<synopsis>
Record to a file.
</synopsis>
<syntax>
<parameter name="filename" required="true" argsep=".">
<argument name="filename" required="true" />
<argument name="format" required="true">
<para>Is the format of the file type to be recorded (wav, gsm, etc).</para>
</argument>
</parameter>
<parameter name="silence">
<para>Is the number of seconds of silence to allow before returning.</para>
</parameter>
<parameter name="maxduration">
<para>Is the maximum recording duration in seconds. If missing
or 0 there is no maximum.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Append to existing recording rather than replacing.</para>
</option>
<option name="n">
<para>Do not answer, but record anyway if line not yet answered.</para>
</option>
<option name="o">
<para>Exit when 0 is pressed, setting the variable <variable>RECORD_STATUS</variable>
to <literal>OPERATOR</literal> instead of <literal>DTMF</literal>
</para>
</option>
<option name="q">
<para>quiet (do not play a beep tone).</para>
</option>
<option name="s">
<para>skip recording if the line is not yet answered.</para>
</option>
<option name="t">
<para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
</option>
<option name="u">
<para>Don't truncate recorded silence.</para>
</option>
<option name="x">
<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
</option>
<option name="k">
<para>Keep recorded file upon hangup.</para>
</option>
<option name="y">
<para>Terminate recording if *any* DTMF digit is received.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>If filename contains <literal>%d</literal>, these characters will be replaced with a number
incremented by one each time the file is recorded.
Use <astcli>core show file formats</astcli> to see the available formats on your system
User can press <literal>#</literal> to terminate the recording and continue to the next priority.
If the user hangs up during a recording, all data will be lost and the application will terminate.</para>
<variablelist>
<variable name="RECORDED_FILE">
<para>Will be set to the final filename of the recording, without an extension.</para>
</variable>
<variable name="RECORD_STATUS">
<para>This is the final status of the command</para>
<value name="DTMF">A terminating DTMF was received ('#' or '*', depending upon option 't')</value>
<value name="SILENCE">The maximum silence occurred in the recording.</value>
<value name="SKIP">The line was not yet answered and the 's' option was specified.</value>
<value name="TIMEOUT">The maximum length was reached.</value>
<value name="HANGUP">The channel was hung up.</value>
<value name="ERROR">An unrecoverable error occurred, which resulted in a WARNING to the logs.</value>
</variable>
</variablelist>
</description>
</application>
<application name="Verbose" language="en_US">
<synopsis>
Send arbitrary text to verbose output.
</synopsis>
<syntax>
<parameter name="level">
<para>Must be an integer value. If not specified, defaults to 0.</para>
</parameter>
<parameter name="message" required="true">
<para>Output text message.</para>
</parameter>
</syntax>
<description>
<para>Sends an arbitrary text message to verbose output.</para>
</description>
</application>
<application name="Log" language="en_US">
<synopsis>
Send arbitrary text to a selected log level.
</synopsis>
<syntax>
<parameter name="level" required="true">
<para>Level must be one of <literal>ERROR</literal>, <literal>WARNING</literal>, <literal>NOTICE</literal>,
<literal>DEBUG</literal>, <literal>VERBOSE</literal> or <literal>DTMF</literal>.</para>
</parameter>
<parameter name="message" required="true">
<para>Output text message.</para>
</parameter>
</syntax>
<description>
<para>Sends an arbitrary text message to a selected log level.</para>
</description>
</application>
<application name="Queue" language="en_US">
<synopsis>
Queue a call for a call queue.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
<parameter name="options">
<optionlist>
<option name="C">
<para>Mark all calls as "answered elsewhere" when cancelled.</para>
</option>
<option name="c">
<para>Continue in the dialplan if the callee hangs up.</para>
</option>
<option name="d">
<para>data-quality (modem) call (minimum delay).</para>
</option>
<option name="F" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>When the caller hangs up, transfer the <emphasis>called member</emphasis>
to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
</option>
<option name="F">
<para>When the caller hangs up, transfer the <emphasis>called member</emphasis> to the next priority of
the current extension and <emphasis>start</emphasis> execution at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
<note>
<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
</note>
</option>
<option name="h">
<para>Allow <emphasis>callee</emphasis> to hang up by pressing <literal>*</literal>.</para>
</option>
<option name="H">
<para>Allow <emphasis>caller</emphasis> to hang up by pressing <literal>*</literal>.</para>
</option>
<option name="n">
<para>No retries on the timeout; will exit this application and
go to the next step.</para>
</option>
<option name="i">
<para>Ignore call forward requests from queue members and do nothing
when they are requested.</para>
</option>
<option name="I">
<para>Asterisk will ignore any connected line update requests or any redirecting party
update requests it may receive on this dial attempt.</para>
</option>
<option name="r">
<para>Ring instead of playing MOH. Periodic Announcements are still made, if applicable.</para>
</option>
<option name="R">
<para>Ring instead of playing MOH when a member channel is actually ringing.</para>
</option>
<option name="t">
<para>Allow the <emphasis>called</emphasis> user to transfer the calling user.</para>
</option>
<option name="T">
<para>Allow the <emphasis>calling</emphasis> user to transfer the call.</para>
</option>
<option name="w">
<para>Allow the <emphasis>called</emphasis> user to write the conversation to
disk via Monitor.</para>
</option>
<option name="W">
<para>Allow the <emphasis>calling</emphasis> user to write the conversation to
disk via Monitor.</para>
</option>
<option name="k">
<para>Allow the <emphasis>called</emphasis> party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="K">
<para>Allow the <emphasis>calling</emphasis> party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="x">
<para>Allow the <emphasis>called</emphasis> user to write the conversation
to disk via MixMonitor.</para>
</option>
<option name="X">
<para>Allow the <emphasis>calling</emphasis> user to write the conversation to
disk via MixMonitor.</para>
</option>
</optionlist>
</parameter>
<parameter name="URL">
<para>
<replaceable>URL</replaceable> will be sent to the called party if the channel supports it.</para>
</parameter>
<parameter name="announceoverride" />
<parameter name="timeout">
<para>Will cause the queue to fail out after a specified number of
seconds, checked between each <filename>queues.conf</filename>
<replaceable>timeout</replaceable> and
<replaceable>retry</replaceable> cycle.</para>
</parameter>
<parameter name="AGI">
<para>Will setup an AGI script to be executed on the calling party's channel once they are
connected to a queue member.</para>
</parameter>
<parameter name="macro">
<para>Will run a macro on the called party's channel (the queue member) once the parties are connected.</para>
</parameter>
<parameter name="gosub">
<para>Will run a gosub on the called party's channel (the queue member) once the parties are connected.</para>
</parameter>
<parameter name="rule">
<para>Will cause the queue's defaultrule to be overridden by the rule specified.</para>
</parameter>
<parameter name="position">
<para>Attempt to enter the caller into the queue at the numerical position specified. <literal>1</literal>
would attempt to enter the caller at the head of the queue, and <literal>3</literal> would attempt to place
the caller third in the queue.</para>
</parameter>
</syntax>
<description>
<para>In addition to transferring the call, a call may be parked and then picked
up by another user.</para>
<para>This application will return to the dialplan if the queue does not exist, or
any of the join options cause the caller to not enter the queue.</para>
<para>This application does not automatically answer and should be preceeded
by an application such as Answer(), Progress(), or Ringing().</para>
<para>This application sets the following channel variables upon completion:</para>
<variablelist>
<variable name="QUEUESTATUS">
<para>The status of the call as a text string.</para>
<value name="TIMEOUT" />
<value name="FULL" />
<value name="JOINEMPTY" />
<value name="LEAVEEMPTY" />
<value name="JOINUNAVAIL" />
<value name="LEAVEUNAVAIL" />
<value name="CONTINUE" />
</variable>
<variable name="ABANDONED">
<para>If the call was not answered by an agent this variable will be TRUE.</para>
<value name="TRUE" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<application name="AddQueueMember" language="en_US">
<synopsis>
Dynamically adds queue members.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
<parameter name="interface" />
<parameter name="penalty" />
<parameter name="options" />
<parameter name="membername" />
<parameter name="stateinterface" />
</syntax>
<description>
<para>Dynamically adds interface to an existing queue. If the interface is
already in the queue it will return an error.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="AQMSTATUS">
<para>The status of the attempt to add a queue member as a text string.</para>
<value name="ADDED" />
<value name="MEMBERALREADY" />
<value name="NOSUCHQUEUE" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<application name="RemoveQueueMember" language="en_US">
<synopsis>
Dynamically removes queue members.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
<parameter name="interface" />
</syntax>
<description>
<para>If the interface is <emphasis>NOT</emphasis> in the queue it will return an error.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="RQMSTATUS">
<value name="REMOVED" />
<value name="NOTINQUEUE" />
<value name="NOSUCHQUEUE" />
<value name="NOTDYNAMIC" />
</variable>
</variablelist>
<para>Example: RemoveQueueMember(techsupport,SIP/3000)</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<application name="PauseQueueMember" language="en_US">
<synopsis>
Pauses a queue member.
</synopsis>
<syntax>
<parameter name="queuename" />
<parameter name="interface" required="true" />
<parameter name="options" />
<parameter name="reason">
<para>Is used to add extra information to the appropriate queue_log entries and manager events.</para>
</parameter>
</syntax>
<description>
<para>Pauses (blocks calls for) a queue member. The given interface will be paused in the given queue.
This prevents any calls from being sent from the queue to the interface until it is
unpaused with UnpauseQueueMember or the manager interface. If no queuename is given,
the interface is paused in every queue it is a member of. The application will fail if the
interface is not found.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PQMSTATUS">
<para>The status of the attempt to pause a queue member as a text string.</para>
<value name="PAUSED" />
<value name="NOTFOUND" />
</variable>
</variablelist>
<para>Example: PauseQueueMember(,SIP/3000)</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<application name="UnpauseQueueMember" language="en_US">
<synopsis>
Unpauses a queue member.
</synopsis>
<syntax>
<parameter name="queuename" />
<parameter name="interface" required="true" />
<parameter name="options" />
<parameter name="reason">
<para>Is used to add extra information to the appropriate queue_log entries and manager events.</para>
</parameter>
</syntax>
<description>
<para>Unpauses (resumes calls to) a queue member. This is the counterpart to <literal>PauseQueueMember()</literal>
and operates exactly the same way, except it unpauses instead of pausing the given interface.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="UPQMSTATUS">
<para>The status of the attempt to unpause a queue member as a text string.</para>
<value name="UNPAUSED" />
<value name="NOTFOUND" />
</variable>
</variablelist>
<para>Example: UnpauseQueueMember(,SIP/3000)</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<application name="QueueLog" language="en_US">
<synopsis>
Writes to the queue_log file.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
<parameter name="uniqueid" required="true" />
<parameter name="agent" required="true" />
<parameter name="event" required="true" />
<parameter name="additionalinfo" />
</syntax>
<description>
<para>Allows you to write your own events into the queue log.</para>
<para>Example: QueueLog(101,${UNIQUEID},${AGENT},WENTONBREAK,600)</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</application>
<function name="QUEUE_VARIABLES" language="en_US">
<synopsis>
Return Queue information in variables.
</synopsis>
<syntax>
<parameter name="queuename" required="true">
<enumlist>
<enum name="QUEUEMAX">
<para>Maxmimum number of calls allowed.</para>
</enum>
<enum name="QUEUESTRATEGY">
<para>The strategy of the queue.</para>
</enum>
<enum name="QUEUECALLS">
<para>Number of calls currently in the queue.</para>
</enum>
<enum name="QUEUEHOLDTIME">
<para>Current average hold time.</para>
</enum>
<enum name="QUEUECOMPLETED">
<para>Number of completed calls for the queue.</para>
</enum>
<enum name="QUEUEABANDONED">
<para>Number of abandoned calls.</para>
</enum>
<enum name="QUEUESRVLEVEL">
<para>Queue service level.</para>
</enum>
<enum name="QUEUESRVLEVELPERF">
<para>Current service level performance.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Makes the following queue variables available.</para>
<para>Returns <literal>0</literal> if queue is found and setqueuevar is defined, <literal>-1</literal> otherwise.</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_MEMBER" language="en_US">
<synopsis>
Count number of members answering a queue.
</synopsis>
<syntax>
<parameter name="queuename" required="false" />
<parameter name="option" required="true">
<enumlist>
<enum name="logged">
<para>Returns the number of logged-in members for the specified queue.</para>
</enum>
<enum name="free">
<para>Returns the number of logged-in members for the specified queue that either can take calls or are currently wrapping up after a previous call.</para>
</enum>
<enum name="ready">
<para>Returns the number of logged-in members for the specified queue that are immediately available to answer a call.</para>
</enum>
<enum name="count">
<para>Returns the total number of members for the specified queue.</para>
</enum>
<enum name="penalty">
<para>Gets or sets queue member penalty. If
<replaceable>queuename</replaceable> is not specified
when setting the penalty then the penalty is set in all queues
the interface is a member.</para>
</enum>
<enum name="paused">
<para>Gets or sets queue member paused status. If
<replaceable>queuename</replaceable> is not specified
when setting the paused status then the paused status is set
in all queues the interface is a member.</para>
</enum>
<enum name="ringinuse">
<para>Gets or sets queue member ringinuse. If
<replaceable>queuename</replaceable> is not specified
when setting ringinuse then ringinuse is set
in all queues the interface is a member.</para>
</enum>
</enumlist>
</parameter>
<parameter name="interface" required="false" />
</syntax>
<description>
<para>Allows access to queue counts [R] and member information [R/W].</para>
<para>
<replaceable>queuename</replaceable> is required for all read operations.</para>
<para>
<replaceable>interface</replaceable> is required for all member operations.</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_MEMBER_COUNT" language="en_US">
<synopsis>
Count number of members answering a queue.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
</syntax>
<description>
<para>Returns the number of members currently associated with the specified <replaceable>queuename</replaceable>.</para>
<warning>
<para>This function has been deprecated in favor of the <literal>QUEUE_MEMBER()</literal> function</para>
</warning>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_EXISTS" language="en_US">
<synopsis>
Check if a named queue exists on this server
</synopsis>
<syntax>
<parameter name="queuename" />
</syntax>
<description>
<para>Returns 1 if the specified queue exists, 0 if it does not</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_WAITING_COUNT" language="en_US">
<synopsis>
Count number of calls currently waiting in a queue.
</synopsis>
<syntax>
<parameter name="queuename" />
</syntax>
<description>
<para>Returns the number of callers currently waiting in the specified <replaceable>queuename</replaceable>.</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_MEMBER_LIST" language="en_US">
<synopsis>
Returns a list of interfaces on a queue.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
</syntax>
<description>
<para>Returns a comma-separated list of members associated with the specified <replaceable>queuename</replaceable>.</para>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<function name="QUEUE_MEMBER_PENALTY" language="en_US">
<synopsis>
Gets or sets queue members penalty.
</synopsis>
<syntax>
<parameter name="queuename" required="true" />
<parameter name="interface" required="true" />
</syntax>
<description>
<para>Gets or sets queue members penalty.</para>
<warning>
<para>This function has been deprecated in favor of the <literal>QUEUE_MEMBER()</literal> function</para>
</warning>
</description>
<see-also>
<ref type="application">Queue</ref>
<ref type="application">QueueLog</ref>
<ref type="application">AddQueueMember</ref>
<ref type="application">RemoveQueueMember</ref>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnpauseQueueMember</ref>
<ref type="function">QUEUE_VARIABLES</ref>
<ref type="function">QUEUE_MEMBER</ref>
<ref type="function">QUEUE_MEMBER_COUNT</ref>
<ref type="function">QUEUE_EXISTS</ref>
<ref type="function">QUEUE_WAITING_COUNT</ref>
<ref type="function">QUEUE_MEMBER_LIST</ref>
<ref type="function">QUEUE_MEMBER_PENALTY</ref>
</see-also>
</function>
<manager name="Queues" language="en_US">
<synopsis>
Queues.
</synopsis>
<syntax>
</syntax>
<description>
<para>Show queues information.</para>
</description>
</manager>
<manager name="QueueStatus" language="en_US">
<synopsis>
Show queue status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue">
<para>Limit the response to the status of the specified queue.</para>
</parameter>
<parameter name="Member">
<para>Limit the response to the status of the specified member.</para>
</parameter>
</syntax>
<description>
<para>Check the status of one or more queues.</para>
</description>
</manager>
<manager name="QueueSummary" language="en_US">
<synopsis>
Show queue summary.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue">
<para>Queue for which the summary is requested.</para>
</parameter>
</syntax>
<description>
<para>Request the manager to send a QueueSummary event.</para>
</description>
</manager>
<manager name="QueueAdd" language="en_US">
<synopsis>
Add interface to queue.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue" required="true">
<para>Queue's name.</para>
</parameter>
<parameter name="Interface" required="true">
<para>The name of the interface (tech/name) to add to the queue.</para>
</parameter>
<parameter name="Penalty">
<para>A penalty (number) to apply to this member. Asterisk will distribute calls to members with higher penalties only after attempting to distribute calls to those with lower penalty.</para>
</parameter>
<parameter name="Paused">
<para>To pause or not the member initially (true/false or 1/0).</para>
</parameter>
<parameter name="MemberName">
<para>Text alias for the interface.</para>
</parameter>
<parameter name="StateInterface" />
</syntax>
<description>
</description>
</manager>
<manager name="QueueRemove" language="en_US">
<synopsis>
Remove interface from queue.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue" required="true">
<para>The name of the queue to take action on.</para>
</parameter>
<parameter name="Interface" required="true">
<para>The interface (tech/name) to remove from queue.</para>
</parameter>
</syntax>
<description>
</description>
</manager>
<manager name="QueuePause" language="en_US">
<synopsis>
Makes a queue member temporarily unavailable.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Interface" required="true">
<para>The name of the interface (tech/name) to pause or unpause.</para>
</parameter>
<parameter name="Paused" required="true">
<para>Pause or unpause the interface. Set to 'true' to pause the member or 'false' to unpause.</para>
</parameter>
<parameter name="Queue">
<para>The name of the queue in which to pause or unpause this member. If not specified, the member will be paused or unpaused in all the queues it is a member of.</para>
</parameter>
<parameter name="Reason">
<para>Text description, returned in the event QueueMemberPaused.</para>
</parameter>
</syntax>
<description>
<para>Pause or unpause a member in a queue.</para>
</description>
</manager>
<manager name="QueueLog" language="en_US">
<synopsis>
Adds custom entry in queue_log.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue" required="true" />
<parameter name="Event" required="true" />
<parameter name="Uniqueid" />
<parameter name="Interface" />
<parameter name="Message" />
</syntax>
<description>
</description>
</manager>
<manager name="QueuePenalty" language="en_US">
<synopsis>
Set the penalty for a queue member.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Interface" required="true">
<para>The interface (tech/name) of the member whose penalty to change.</para>
</parameter>
<parameter name="Penalty" required="true">
<para>The new penalty (number) for the member. Must be nonnegative.</para>
</parameter>
<parameter name="Queue">
<para>If specified, only set the penalty for the member of this queue. Otherwise, set the penalty for the member in all queues to which the member belongs.</para>
</parameter>
</syntax>
<description>
<para>Change the penalty of a queue member</para>
</description>
</manager>
<manager name="QueueMemberRingInUse" language="en_US">
<synopsis>
Set the ringinuse value for a queue member.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Interface" required="true" />
<parameter name="RingInUse" required="true" />
<parameter name="Queue" />
</syntax>
<description>
</description>
</manager>
<manager name="QueueRule" language="en_US">
<synopsis>
Queue Rules.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Rule">
<para>The name of the rule in queuerules.conf whose contents to list.</para>
</parameter>
</syntax>
<description>
<para>List queue rules defined in queuerules.conf</para>
</description>
</manager>
<manager name="QueueReload" language="en_US">
<synopsis>
Reload a queue, queues, or any sub-section of a queue or queues.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue">
<para>The name of the queue to take action on. If no queue name is specified, then all queues are affected.</para>
</parameter>
<parameter name="Members">
<para>Whether to reload the queue's members.</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
</enumlist>
</parameter>
<parameter name="Rules">
<para>Whether to reload queuerules.conf</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
</enumlist>
</parameter>
<parameter name="Parameters">
<para>Whether to reload the other queue options.</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
</enumlist>
</parameter>
</syntax>
<description>
</description>
</manager>
<manager name="QueueReset" language="en_US">
<synopsis>
Reset queue statistics.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Queue">
<para>The name of the queue on which to reset statistics.</para>
</parameter>
</syntax>
<description>
<para>Reset the statistics for a queue.</para>
</description>
</manager>
<managerEvent language="en_US" name="QueueMemberStatus">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a Queue member's status has changed.</synopsis>
<syntax>
<parameter name="Queue">
<para>The name of the queue.</para>
</parameter>
<parameter name="MemberName">
<para>The name of the queue member.</para>
</parameter>
<parameter name="Interface">
<para>The queue member's channel technology or location.</para>
</parameter>
<parameter name="StateInterface">
<para>Channel technology or location from which to read device state changes.</para>
</parameter>
<parameter name="Membership">
<enumlist>
<enum name="dynamic"/>
<enum name="realtime"/>
<enum name="static"/>
</enumlist>
</parameter>
<parameter name="Penalty">
<para>The penalty associated with the queue member.</para>
</parameter>
<parameter name="CallsTaken">
<para>The number of calls this queue member has serviced.</para>
</parameter>
<parameter name="LastCall">
<para>The time this member last took a call, expressed in seconds since 00:00, Jan 1, 1970 UTC.</para>
</parameter>
<parameter name="InCall">
<para>Set to 1 if member is in call. Set to 0 after LastCall time is updated.</para>
<enumlist>
<enum name="0"/>
<enum name="1"/>
</enumlist>
</parameter>
<parameter name="Status">
<para>The numeric device state status of the queue member.</para>
<enumlist>
<enum name="0">
<para>AST_DEVICE_UNKNOWN</para>
</enum>
<enum name="1">
<para>AST_DEVICE_NOT_INUSE</para>
</enum>
<enum name="2">
<para>AST_DEVICE_INUSE</para>
</enum>
<enum name="3">
<para>AST_DEVICE_BUSY</para>
</enum>
<enum name="4">
<para>AST_DEVICE_INVALID</para>
</enum>
<enum name="5">
<para>AST_DEVICE_UNAVAILABLE</para>
</enum>
<enum name="6">
<para>AST_DEVICE_RINGING</para>
</enum>
<enum name="7">
<para>AST_DEVICE_RINGINUSE</para>
</enum>
<enum name="8">
<para>AST_DEVICE_ONHOLD</para>
</enum>
</enumlist>
</parameter>
<parameter name="Paused">
<enumlist>
<enum name="0"/>
<enum name="1"/>
</enumlist>
</parameter>
<parameter name="PausedReason">
<para>If set when paused, the reason the queue member was paused.</para>
</parameter>
<parameter name="Ringinuse">
<enumlist>
<enum name="0"/>
<enum name="1"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueMemberAdded">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a member is added to the queue.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="managerEvent">QueueMemberRemoved</ref>
<ref type="application">AddQueueMember</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueMemberRemoved">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a member is removed from the queue.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="managerEvent">QueueMemberAdded</ref>
<ref type="application">RemoveQueueMember</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueMemberPause">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a member is paused/unpaused in the queue.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter)" />
<parameter name="Reason">
<para>The reason a member was paused.</para>
</parameter>
</syntax>
<see-also>
<ref type="application">PauseQueueMember</ref>
<ref type="application">UnPauseQueueMember</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueMemberPenalty">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a member's penalty is changed.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="function">QUEUE_MEMBER</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueMemberRinginuse">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a member's ringinuse setting is changed.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="function">QUEUE_MEMBER</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueCallerJoin">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a caller joins a Queue.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<parameter name="Position">
<para>This channel's current position in the queue.</para>
</parameter>
<parameter name="Count">
<para>The total number of channels in the queue.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">QueueCallerLeave</ref>
<ref type="application">Queue</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueCallerLeave">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a caller leaves a Queue.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueCallerJoin']/managerEventInstance/syntax/parameter[@name='Count'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueCallerJoin']/managerEventInstance/syntax/parameter[@name='Position'])" />
</syntax>
<see-also>
<ref type="managerEvent">QueueCallerJoin</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="QueueCallerAbandon">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a caller abandons the queue.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueCallerJoin']/managerEventInstance/syntax/parameter[@name='Position'])" />
<parameter name="OriginalPosition">
<para>The channel's original position in the queue.</para>
</parameter>
<parameter name="HoldTime">
<para>The time the channel was in the queue, expressed in seconds since 00:00, Jan 1, 1970 UTC.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentCalled">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when an queue member is notified of a caller in the queue.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Interface'])" />
</syntax>
<see-also>
<ref type="managerEvent">AgentRingNoAnswer</ref>
<ref type="managerEvent">AgentComplete</ref>
<ref type="managerEvent">AgentConnect</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentRingNoAnswer">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a queue member is notified of a caller in the queue and fails to answer.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Interface'])" />
<parameter name="RingTime">
<para>The time the queue member was rung, expressed in seconds since 00:00, Jan 1, 1970 UTC.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AgentCalled</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentComplete">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a queue member has finished servicing a caller in the queue.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Interface'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueCallerAbandon']/managerEventInstance/syntax/parameter[@name='HoldTime'])" />
<parameter name="TalkTime">
<para>The time the queue member talked with the caller in the queue, expressed in seconds since 00:00, Jan 1, 1970 UTC.</para>
</parameter>
<parameter name="Reason">
<enumlist>
<enum name="caller"/>
<enum name="agent"/>
<enum name="transfer"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AgentCalled</ref>
<ref type="managerEvent">AgentConnect</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentDump">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a queue member hangs up on a caller in the queue.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Interface'])" />
</syntax>
<see-also>
<ref type="managerEvent">AgentCalled</ref>
<ref type="managerEvent">AgentConnect</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentConnect">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when a queue member answers and is bridged to a caller in the queue.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Interface'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AgentRingNoAnswer']/managerEventInstance/syntax/parameter[@name='RingTime'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueCallerAbandon']/managerEventInstance/syntax/parameter[@name='HoldTime'])" />
</syntax>
<see-also>
<ref type="managerEvent">AgentCalled</ref>
<ref type="managerEvent">AgentComplete</ref>
<ref type="managerEvent">AgentDump</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<application name="OSPAuth" language="en_US">
<synopsis>
OSP Authentication.
</synopsis>
<syntax>
<parameter name="provider">
<para>The name of the provider that authenticates the call.</para>
</parameter>
<parameter name="options">
<para>Reserverd.</para>
</parameter>
</syntax>
<description>
<para>Authenticate a call by OSP.</para>
<para>Input variables:</para>
<variablelist>
<variable name="OSPINPEERIP">
<para>The last hop IP address.</para>
</variable>
<variable name="OSPINTOKEN">
<para>The inbound OSP token.</para>
</variable>
</variablelist>
<para>Output variables:</para>
<variablelist>
<variable name="OSPINHANDLE">
<para>The inbound call OSP transaction handle.</para>
</variable>
<variable name="OSPINTIMELIMIT">
<para>The inbound call duration limit in seconds.</para>
</variable>
</variablelist>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="OSPAUTHSTATUS">
<para>The status of OSPAuth attempt as a text string, one of</para>
<value name="SUCCESS" />
<value name="FAILED" />
<value name="ERROR" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">OSPLookup</ref>
<ref type="application">OSPNext</ref>
<ref type="application">OSPFinish</ref>
</see-also>
</application>
<application name="OSPLookup" language="en_US">
<synopsis>
Lookup destination by OSP.
</synopsis>
<syntax>
<parameter name="exten" required="true">
<para>The exten of the call.</para>
</parameter>
<parameter name="provider">
<para>The name of the provider that is used to route the call.</para>
</parameter>
<parameter name="options">
<enumlist>
<enum name="h">
<para>generate H323 call id for the outbound call</para>
</enum>
<enum name="s">
<para>generate SIP call id for the outbound call. Have not been implemented</para>
</enum>
<enum name="i">
<para>generate IAX call id for the outbound call. Have not been implemented</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Looks up destination via OSP.</para>
<para>Input variables:</para>
<variablelist>
<variable name="OSPINACTUALSRC">
<para>The actual source device IP address in indirect mode.</para>
</variable>
<variable name="OSPINPEERIP">
<para>The last hop IP address.</para>
</variable>
<variable name="OSPINTECH">
<para>The inbound channel technology for the call.</para>
</variable>
<variable name="OSPINHANDLE">
<para>The inbound call OSP transaction handle.</para>
</variable>
<variable name="OSPINTIMELIMIT">
<para>The inbound call duration limit in seconds.</para>
</variable>
<variable name="OSPINNETWORKID">
<para>The inbound source network ID.</para>
</variable>
<variable name="OSPINNPRN">
<para>The inbound routing number.</para>
</variable>
<variable name="OSPINNPCIC">
<para>The inbound carrier identification code.</para>
</variable>
<variable name="OSPINNPDI">
<para>The inbound number portability database dip indicator.</para>
</variable>
<variable name="OSPINSPID">
<para>The inbound service provider identity.</para>
</variable>
<variable name="OSPINOCN">
<para>The inbound operator company number.</para>
</variable>
<variable name="OSPINSPN">
<para>The inbound service provider name.</para>
</variable>
<variable name="OSPINALTSPN">
<para>The inbound alternate service provider name.</para>
</variable>
<variable name="OSPINMCC">
<para>The inbound mobile country code.</para>
</variable>
<variable name="OSPINMNC">
<para>The inbound mobile network code.</para>
</variable>
<variable name="OSPINTOHOST">
<para>The inbound To header host part.</para>
</variable>
<variable name="OSPINRPIDUSER">
<para>The inbound Remote-Party-ID header user part.</para>
</variable>
<variable name="OSPINPAIUSER">
<para>The inbound P-Asserted-Identify header user part.</para>
</variable>
<variable name="OSPINDIVUSER">
<para>The inbound Diversion header user part.</para>
</variable>
<variable name="OSPINDIVHOST">
<para>The inbound Diversion header host part.</para>
</variable>
<variable name="OSPINPCIUSER">
<para>The inbound P-Charge-Info header user part.</para>
</variable>
<variable name="OSPINCUSTOMINFOn">
<para>The inbound custom information, where <literal>n</literal> is the index beginning with <literal>1</literal>
upto <literal>8</literal>.</para>
</variable>
</variablelist>
<para>Output variables:</para>
<variablelist>
<variable name="OSPOUTHANDLE">
<para>The outbound call OSP transaction handle.</para>
</variable>
<variable name="OSPOUTTECH">
<para>The outbound channel technology for the call.</para>
</variable>
<variable name="OSPDESTINATION">
<para>The outbound destination IP address.</para>
</variable>
<variable name="OSPOUTCALLING">
<para>The outbound calling number.</para>
</variable>
<variable name="OSPOUTCALLED">
<para>The outbound called number.</para>
</variable>
<variable name="OSPOUTNETWORKID">
<para>The outbound destination network ID.</para>
</variable>
<variable name="OSPOUTNPRN">
<para>The outbound routing number.</para>
</variable>
<variable name="OSPOUTNPCIC">
<para>The outbound carrier identification code.</para>
</variable>
<variable name="OSPOUTNPDI">
<para>The outbound number portability database dip indicator.</para>
</variable>
<variable name="OSPOUTSPID">
<para>The outbound service provider identity.</para>
</variable>
<variable name="OSPOUTOCN">
<para>The outbound operator company number.</para>
</variable>
<variable name="OSPOUTSPN">
<para>The outbound service provider name.</para>
</variable>
<variable name="OSPOUTALTSPN">
<para>The outbound alternate service provider name.</para>
</variable>
<variable name="OSPOUTMCC">
<para>The outbound mobile country code.</para>
</variable>
<variable name="OSPOUTMNC">
<para>The outbound mobile network code.</para>
</variable>
<variable name="OSPOUTTOKEN">
<para>The outbound OSP token.</para>
</variable>
<variable name="OSPDESTREMAILS">
<para>The number of remained destinations.</para>
</variable>
<variable name="OSPOUTTIMELIMIT">
<para>The outbound call duration limit in seconds.</para>
</variable>
<variable name="OSPOUTCALLIDTYPES">
<para>The outbound Call-ID types.</para>
</variable>
<variable name="OSPOUTCALLID">
<para>The outbound Call-ID. Only for H.323.</para>
</variable>
<variable name="OSPDIALSTR">
<para>The outbound Dial command string.</para>
</variable>
</variablelist>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="OSPLOOKUPSTATUS">
<para>The status of OSPLookup attempt as a text string, one of</para>
<value name="SUCCESS" />
<value name="FAILED" />
<value name="ERROR" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">OSPAuth</ref>
<ref type="application">OSPNext</ref>
<ref type="application">OSPFinish</ref>
</see-also>
</application>
<application name="OSPNext" language="en_US">
<synopsis>
Lookup next destination by OSP.
</synopsis>
<description>
<para>Looks up the next destination via OSP.</para>
<para>Input variables:</para>
<variablelist>
<variable name="OSPINHANDLE">
<para>The inbound call OSP transaction handle.</para>
</variable>
<variable name="OSPOUTHANDLE">
<para>The outbound call OSP transaction handle.</para>
</variable>
<variable name="OSPINTIMELIMIT">
<para>The inbound call duration limit in seconds.</para>
</variable>
<variable name="OSPOUTCALLIDTYPES">
<para>The outbound Call-ID types.</para>
</variable>
<variable name="OSPDESTREMAILS">
<para>The number of remained destinations.</para>
</variable>
</variablelist>
<para>Output variables:</para>
<variablelist>
<variable name="OSPOUTTECH">
<para>The outbound channel technology.</para>
</variable>
<variable name="OSPDESTINATION">
<para>The destination IP address.</para>
</variable>
<variable name="OSPOUTCALLING">
<para>The outbound calling number.</para>
</variable>
<variable name="OSPOUTCALLED">
<para>The outbound called number.</para>
</variable>
<variable name="OSPOUTNETWORKID">
<para>The outbound destination network ID.</para>
</variable>
<variable name="OSPOUTNPRN">
<para>The outbound routing number.</para>
</variable>
<variable name="OSPOUTNPCIC">
<para>The outbound carrier identification code.</para>
</variable>
<variable name="OSPOUTNPDI">
<para>The outbound number portability database dip indicator.</para>
</variable>
<variable name="OSPOUTSPID">
<para>The outbound service provider identity.</para>
</variable>
<variable name="OSPOUTOCN">
<para>The outbound operator company number.</para>
</variable>
<variable name="OSPOUTSPN">
<para>The outbound service provider name.</para>
</variable>
<variable name="OSPOUTALTSPN">
<para>The outbound alternate service provider name.</para>
</variable>
<variable name="OSPOUTMCC">
<para>The outbound mobile country code.</para>
</variable>
<variable name="OSPOUTMNC">
<para>The outbound mobile network code.</para>
</variable>
<variable name="OSPOUTTOKEN">
<para>The outbound OSP token.</para>
</variable>
<variable name="OSPDESTREMAILS">
<para>The number of remained destinations.</para>
</variable>
<variable name="OSPOUTTIMELIMIT">
<para>The outbound call duration limit in seconds.</para>
</variable>
<variable name="OSPOUTCALLID">
<para>The outbound Call-ID. Only for H.323.</para>
</variable>
<variable name="OSPDIALSTR">
<para>The outbound Dial command string.</para>
</variable>
</variablelist>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="OSPNEXTSTATUS">
<para>The status of the OSPNext attempt as a text string, one of</para>
<value name="SUCCESS" />
<value name="FAILED" />
<value name="ERROR" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">OSPAuth</ref>
<ref type="application">OSPLookup</ref>
<ref type="application">OSPFinish</ref>
</see-also>
</application>
<application name="OSPFinish" language="en_US">
<synopsis>
Report OSP entry.
</synopsis>
<syntax>
<parameter name="cause">
<para>Hangup cause.</para>
</parameter>
<parameter name="options">
<para>Reserved.</para>
</parameter>
</syntax>
<description>
<para>Report call state.</para>
<para>Input variables:</para>
<variablelist>
<variable name="OSPINHANDLE">
<para>The inbound call OSP transaction handle.</para>
</variable>
<variable name="OSPOUTHANDLE">
<para>The outbound call OSP transaction handle.</para>
</variable>
<variable name="OSPAUTHSTATUS">
<para>The OSPAuth status.</para>
</variable>
<variable name="OSPLOOKUPSTATUS">
<para>The OSPLookup status.</para>
</variable>
<variable name="OSPNEXTSTATUS">
<para>The OSPNext status.</para>
</variable>
<variable name="OSPINAUDIOQOS">
<para>The inbound call leg audio QoS string.</para>
</variable>
<variable name="OSPOUTAUDIOQOS">
<para>The outbound call leg audio QoS string.</para>
</variable>
</variablelist>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="OSPFINISHSTATUS">
<para>The status of the OSPFinish attempt as a text string, one of</para>
<value name="SUCCESS" />
<value name="FAILED" />
<value name="ERROR" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">OSPAuth</ref>
<ref type="application">OSPLookup</ref>
<ref type="application">OSPNext</ref>
</see-also>
</application>
<application name="SendURL" language="en_US">
<synopsis>
Send a URL.
</synopsis>
<syntax>
<parameter name="URL" required="true" />
<parameter name="option">
<optionlist>
<option name="w">
<para>Execution will wait for an acknowledgement that the
URL has been loaded before continuing.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Requests client go to <replaceable>URL</replaceable> (IAX2) or sends the
URL to the client (other channels).</para>
<para>Result is returned in the <variable>SENDURLSTATUS</variable> channel variable:</para>
<variablelist>
<variable name="SENDURLSTATUS">
<value name="SUCCESS">
URL successfully sent to client.
</value>
<value name="FAILURE">
Failed to send URL.
</value>
<value name="NOLOAD">
Client failed to load URL (wait enabled).
</value>
<value name="UNSUPPORTED">
Channel does not support URL transport.
</value>
</variable>
</variablelist>
<para>SendURL continues normally if the URL was sent correctly or if the channel
does not support HTML transport. Otherwise, the channel is hung up.</para>
</description>
<see-also>
<ref type="application">SendImage</ref>
<ref type="application">SendText</ref>
</see-also>
</application>
<application name="SetCallerPres" language="en_US">
<synopsis>
Set CallerID Presentation.
</synopsis>
<syntax>
<parameter name="presentation" required="true">
<enumlist>
<enum name="allowed_not_screened">
<para>Presentation Allowed, Not Screened.</para>
</enum>
<enum name="allowed_passed_screen">
<para>Presentation Allowed, Passed Screen.</para>
</enum>
<enum name="allowed_failed_screen">
<para>Presentation Allowed, Failed Screen.</para>
</enum>
<enum name="allowed">
<para>Presentation Allowed, Network Number.</para>
</enum>
<enum name="prohib_not_screened">
<para>Presentation Prohibited, Not Screened.</para>
</enum>
<enum name="prohib_passed_screen">
<para>Presentation Prohibited, Passed Screen.</para>
</enum>
<enum name="prohib_failed_screen">
<para>Presentation Prohibited, Failed Screen.</para>
</enum>
<enum name="prohib">
<para>Presentation Prohibited, Network Number.</para>
</enum>
<enum name="unavailable">
<para>Number Unavailable.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Set Caller*ID presentation on a call.</para>
</description>
</application>
<application name="ADSIProg" language="en_US">
<synopsis>
Load Asterisk ADSI Scripts into phone
</synopsis>
<syntax>
<parameter name="script" required="false">
<para>adsi script to use. If not given uses the default script <filename>asterisk.adsi</filename></para>
</parameter>
</syntax>
<description>
<para>This application programs an ADSI Phone with the given script</para>
</description>
<see-also>
<ref type="application">GetCPEID</ref>
<ref type="filename">adsi.conf</ref>
</see-also>
</application>
<application name="Stasis" language="en_US">
<synopsis>Invoke an external Stasis application.</synopsis>
<syntax>
<parameter name="app_name" required="true">
<para>Name of the application to invoke.</para>
</parameter>
<parameter name="args">
<para>Optional comma-delimited arguments for the
application invocation.</para>
</parameter>
</syntax>
<description>
<para>Invoke a Stasis application.</para>
<para>This application will set the following channel variable upon
completion:</para>
<variablelist>
<variable name="STASISSTATUS">
<para>This indicates the status of the execution of the
Stasis application.</para>
<value name="SUCCESS">
The channel has exited Stasis without any failures in
Stasis.
</value>
<value name="FAILED">
A failure occurred when executing the Stasis
The app registry is not instantiated; The app
application. Some (not all) possible reasons for this:
requested is not registered; The app requested is not
active; Stasis couldn't send a start message.
</value>
</variable>
</variablelist>
</description>
</application>
<configInfo name="app_confbridge" language="en_US">
<synopsis>Conference Bridge Application</synopsis>
<configFile name="confbridge.conf">
<configObject name="global">
<synopsis>Unused, but reserved.</synopsis>
</configObject>
<configObject name="user_profile">
<synopsis>A named profile to apply to specific callers.</synopsis>
<description>
<para>Callers in a ConfBridge have a profile associated with them
that determine their options. A configuration section is determined to be a
user_profile when the <literal>type</literal> parameter has a value
of <literal>user</literal>.
</para>
</description>
<configOption name="type">
<synopsis>Define this configuration category as a user profile.</synopsis>
<description>
<para>The type parameter determines how a context in the
configuration file is interpreted.</para>
<enumlist>
<enum name="user">
<para>Configure the context as a <replaceable>user_profile</replaceable></para>
</enum>
<enum name="bridge">
<para>Configure the context as a <replaceable>bridge_profile</replaceable></para>
</enum>
<enum name="menu">
<para>Configure the context as a <replaceable>menu</replaceable></para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="admin">
<synopsis>Sets if the user is an admin or not</synopsis>
</configOption>
<configOption name="marked">
<synopsis>Sets if this is a marked user or not</synopsis>
</configOption>
<configOption name="startmuted">
<synopsis>Sets if all users should start out muted</synopsis>
</configOption>
<configOption name="music_on_hold_when_empty">
<synopsis>Play MOH when user is alone or waiting on a marked user</synopsis>
</configOption>
<configOption name="quiet">
<synopsis>Silence enter/leave prompts and user intros for this user</synopsis>
</configOption>
<configOption name="announce_user_count">
<synopsis>Sets if the number of users should be announced to the user</synopsis>
</configOption>
<configOption name="announce_user_count_all">
<synopsis>Announce user count to all the other users when this user joins</synopsis>
<description>
<para>Sets if the number of users should be announced to all the other users
in the conference when this user joins. This option can be either set to 'yes' or
a number. When set to a number, the announcement will only occur once the user
count is above the specified number.
</para>
</description>
</configOption>
<configOption name="announce_only_user">
<synopsis>Announce to a user when they join an empty conference</synopsis>
</configOption>
<configOption name="wait_marked">
<synopsis>Sets if the user must wait for a marked user to enter before joining a conference</synopsis>
</configOption>
<configOption name="end_marked">
<synopsis>Kick the user from the conference when the last marked user leaves</synopsis>
</configOption>
<configOption name="talk_detection_events">
<synopsis>Set whether or not notifications of when a user begins and ends talking should be sent out as events over AMI</synopsis>
</configOption>
<configOption name="dtmf_passthrough">
<synopsis>Sets whether or not DTMF should pass through the conference</synopsis>
</configOption>
<configOption name="announce_join_leave">
<synopsis>Prompt user for their name when joining a conference and play it to the conference when they enter</synopsis>
</configOption>
<configOption name="announce_join_leave_review">
<synopsis>Prompt user for their name when joining a conference and play it to the conference when they enter.
The user will be asked to review the recording of their name before entering the conference.</synopsis>
</configOption>
<configOption name="pin">
<synopsis>Sets a PIN the user must enter before joining the conference</synopsis>
</configOption>
<configOption name="music_on_hold_class">
<synopsis>The MOH class to use for this user</synopsis>
</configOption>
<configOption name="announcement">
<synopsis>Sound file to play to the user when they join a conference</synopsis>
</configOption>
<configOption name="denoise">
<synopsis>Apply a denoise filter to the audio before mixing</synopsis>
<description>
<para>Sets whether or not a denoise filter should be applied
to the audio before mixing or not. Off by default. Requires
<literal>codec_speex</literal> to be built and installed. Do not confuse this option
with <replaceable>drop_silence</replaceable>. Denoise is useful if there is a lot of background
noise for a user as it attempts to remove the noise while preserving
the speech. This option does NOT remove silence from being mixed into
the conference and does come at the cost of a slight performance hit.
</para>
</description>
</configOption>
<configOption name="dsp_drop_silence">
<synopsis>Drop what Asterisk detects as silence from audio sent to the bridge</synopsis>
<description>
<para>
This option drops what Asterisk detects as silence from
entering into the bridge. Enabling this option will drastically
improve performance and help remove the buildup of background
noise from the conference. Highly recommended for large conferences
due to its performance enhancements.
</para>
</description>
</configOption>
<configOption name="dsp_silence_threshold">
<synopsis>The number of milliseconds of detected silence necessary to trigger silence detection</synopsis>
<description>
<para>
The time in milliseconds of sound falling within the what
the dsp has established as baseline silence before a user
is considered be silent. This value affects several
operations and should not be changed unless the impact
on call quality is fully understood.</para>
<para>What this value affects internally:</para>
<para>
1. When talk detection AMI events are enabled, this value
determines when the user has stopped talking after a
period of talking. If this value is set too low
AMI events indicating the user has stopped talking
may get falsely sent out when the user briefly pauses
during mid sentence.
</para>
<para>
2. The <replaceable>drop_silence</replaceable> option depends on this value to
determine when the user's audio should begin to be
dropped from the conference bridge after the user
stops talking. If this value is set too low the user's
audio stream may sound choppy to the other participants.
This is caused by the user transitioning constantly from
silence to talking during mid sentence.
</para>
<para>
The best way to approach this option is to set it slightly above
the maximum amount of ms of silence a user may generate during
natural speech.
</para>
<para>By default this value is 2500ms. Valid values are 1 through 2^31.</para>
</description>
</configOption>
<configOption name="dsp_talking_threshold">
<synopsis>The number of milliseconds of detected non-silence necessary to triger talk detection</synopsis>
<description>
<para>
The time in milliseconds of sound above what the dsp has
established as base line silence for a user before a user
is considered to be talking. This value affects several
operations and should not be changed unless the impact on
call quality is fully understood.</para>
<para>
What this value affects internally:
</para>
<para>
1. Audio is only mixed out of a user's incoming audio stream
if talking is detected. If this value is set too
loose the user will hear themselves briefly each
time they begin talking until the dsp has time to
establish that they are in fact talking.
</para>
<para>
2. When talk detection AMI events are enabled, this value
determines when talking has begun which results in
an AMI event to fire. If this value is set too tight
AMI events may be falsely triggered by variants in
room noise.
</para>
<para>
3. The <replaceable>drop_silence</replaceable> option depends on this value to determine
when the user's audio should be mixed into the bridge
after periods of silence. If this value is too loose
the beginning of a user's speech will get cut off as they
transition from silence to talking.
</para>
<para>By default this value is 160 ms. Valid values are 1 through 2^31</para>
</description>
</configOption>
<configOption name="jitterbuffer">
<synopsis>Place a jitter buffer on the user's audio stream before audio mixing is performed</synopsis>
<description>
<para>
Enabling this option places a jitterbuffer on the user's audio stream
before audio mixing is performed. This is highly recommended but will
add a slight delay to the audio. This option is using the <literal>JITTERBUFFER</literal>
dialplan function's default adaptive jitterbuffer. For a more fine tuned
jitterbuffer, disable this option and use the <literal>JITTERBUFFER</literal> dialplan function
on the user before entering the ConfBridge application.
</para>
</description>
</configOption>
<configOption name="template">
<synopsis>When using the CONFBRIDGE dialplan function, use a user profile as a template for creating a new temporary profile</synopsis>
</configOption>
<configOption name="timeout">
<synopsis>Kick the user out of the conference after this many seconds. 0 means there is no timeout for the user.</synopsis>
</configOption>
</configObject>
<configObject name="bridge_profile">
<synopsis>A named profile to apply to specific bridges.</synopsis>
<description>
<para>ConfBridge bridges have a profile associated with them
that determine their options. A configuration section is determined to be a
<literal>bridge_profile</literal> when the <literal>type</literal> parameter has a value
of <literal>bridge</literal>.
</para>
</description>
<configOption name="type">
<synopsis>Define this configuration category as a bridge profile</synopsis>
<description>
<para>The type parameter determines how a context in the
configuration file is interpreted.</para>
<enumlist>
<enum name="user">
<para>Configure the context as a <replaceable>user_profile</replaceable></para>
</enum>
<enum name="bridge">
<para>Configure the context as a <replaceable>bridge_profile</replaceable></para>
</enum>
<enum name="menu">
<para>Configure the context as a <replaceable>menu</replaceable></para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="jitterbuffer">
<synopsis>Place a jitter buffer on the conference's audio stream</synopsis>
</configOption>
<configOption name="internal_sample_rate">
<synopsis>Set the internal native sample rate for mixing the conference</synopsis>
<description>
<para>
Sets the internal native sample rate the
conference is mixed at. This is set to automatically
adjust the sample rate to the best quality by default.
Other values can be anything from 8000-192000. If a
sample rate is set that Asterisk does not support, the
closest sample rate Asterisk does support to the one requested
will be used.
</para>
</description>
</configOption>
<configOption name="language" default="en">
<synopsis>The language used for announcements to the conference.</synopsis>
<description>
<para>
By default, announcements to a conference use English. Which means
the prompts played to all users within the conference will be
English. By changing the language of a bridge, this will change
the language of the prompts played to all users.
</para>
</description>
</configOption>
<configOption name="mixing_interval">
<synopsis>Sets the internal mixing interval in milliseconds for the bridge</synopsis>
<description>
<para>
Sets the internal mixing interval in milliseconds for the bridge. This
number reflects how tight or loose the mixing will be for the conference.
In order to improve performance a larger mixing interval such as 40ms may
be chosen. Using a larger mixing interval comes at the cost of introducing
larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
or 80.
</para>
</description>
</configOption>
<configOption name="record_conference">
<synopsis>Record the conference starting with the first active user's entrance and ending with the last active user's exit</synopsis>
<description>
<para>
Records the conference call starting when the first user
enters the room, and ending when the last user exits the room.
The default recorded filename is
<filename>'confbridge-${name of conference bridge}-${start time}.wav'</filename>
and the default format is 8khz slinear. This file will be
located in the configured monitoring directory in <filename>asterisk.conf</filename>.
</para>
</description>
</configOption>
<configOption name="record_file" default="confbridge-${name of conference bridge}-${start time}.wav">
<synopsis>The filename of the conference recording</synopsis>
<description>
<para>
When <replaceable>record_conference</replaceable> is set to yes, the specific name of the
record file can be set using this option. Note that since multiple
conferences may use the same bridge profile, this may cause issues
depending on the configuration. It is recommended to only use this
option dynamically with the <literal>CONFBRIDGE()</literal> dialplan function. This
allows the record name to be specified and a unique name to be chosen.
By default, the record_file is stored in Asterisk's spool/monitor directory
with a unique filename starting with the 'confbridge' prefix.
</para>
</description>
</configOption>
<configOption name="record_file_append" default="yes">
<synopsis>Append record file when starting/stopping on same conference recording</synopsis>
<description>
<para>
When <replaceable>record_file_append</replaceable> is set to yes, stopping and starting recording on a
conference adds the new portion to end of current record_file. When this is
set to no, a new <replaceable>record_file</replaceable> is generated every time you start then stop recording
on a conference.
</para>
</description>
</configOption>
<configOption name="regcontext">
<synopsis>The name of the context into which to register the name of the conference bridge as NoOP() at priority 1</synopsis>
<description>
<para>
When set this will cause the name of the created conference to be registered
into the named context at priority 1 with an operation of NoOP(). This can
then be used in other parts of the dialplan to test for the existence of a
specific conference bridge.
You should be aware that there are potential races between testing for the
existence of a bridge, and taking action upon that information, consider
for example two callers executing the check simultaniously, and then taking
special action as "first caller" into the bridge. The same for exiting,
directly after the check the bridge can be destroyed before the new caller
enters (creating a new bridge), for example, and the "first member" actions
could thus be missed.
</para>
</description>
</configOption>
<configOption name="video_mode">
<synopsis>Sets how confbridge handles video distribution to the conference participants</synopsis>
<description>
<para>
Sets how confbridge handles video distribution to the conference participants.
Note that participants wanting to view and be the source of a video feed
<emphasis>MUST</emphasis> be sharing the same video codec. Also, using video in conjunction with
with the jitterbuffer currently results in the audio being slightly out of sync
with the video. This is a result of the jitterbuffer only working on the audio
stream. It is recommended to disable the jitterbuffer when video is used.</para>
<enumlist>
<enum name="none">
<para>No video sources are set by default in the conference. It is still
possible for a user to be set as a video source via AMI or DTMF action
at any time.</para>
</enum>
<enum name="follow_talker">
<para>The video feed will follow whoever is talking and providing video.</para>
</enum>
<enum name="last_marked">
<para>The last marked user to join the conference with video capabilities
will be the single source of video distributed to all participants.
If multiple marked users are capable of video, the last one to join
is always the source, when that user leaves it goes to the one who
joined before them.</para>
</enum>
<enum name="first_marked">
<para>The first marked user to join the conference with video capabilities
is the single source of video distribution among all participants. If
that user leaves, the marked user to join after them becomes the source.</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="max_members">
<synopsis>Limit the maximum number of participants for a single conference</synopsis>
<description>
<para>
This option limits the number of participants for a single
conference to a specific number. By default conferences
have no participant limit. After the limit is reached, the
conference will be locked until someone leaves. Note however
that an Admin user will always be alowed to join the conference
regardless if this limit is reached or not.
</para>
</description>
</configOption>
<configOption name="^sound_">
<synopsis>Override the various conference bridge sound files</synopsis>
<description>
<para>
All sounds in the conference are customizable using the bridge profile options below.
Simply state the option followed by the filename or full path of the filename after
the option. Example: <literal>sound_had_joined=conf-hasjoin</literal> This will play the <literal>conf-hasjoin</literal>
sound file found in the sounds directory when announcing someone's name is joining the
conference.</para>
<enumlist>
<enum name="sound_join">
<para>The sound played to everyone when someone enters the conference.</para>
</enum>
<enum name="sound_leave">
<para>The sound played to everyone when someone leaves the conference.</para>
</enum>
<enum name="sound_has_joined">
<para>The sound played before announcing someone's name has
joined the conference. This is used for user intros.
Example <literal>"_____ has joined the conference"</literal>
</para>
</enum>
<enum name="sound_has_left">
<para>The sound played when announcing someone's name has
left the conference. This is used for user intros.
Example <literal>"_____ has left the conference"</literal>
</para>
</enum>
<enum name="sound_kicked">
<para>The sound played to a user who has been kicked from the conference.</para>
</enum>
<enum name="sound_muted">
<para>The sound played when the mute option it toggled on.</para>
</enum>
<enum name="sound_unmuted">
<para>The sound played when the mute option it toggled off.</para>
</enum>
<enum name="sound_only_person">
<para>The sound played when the user is the only person in the conference.</para>
</enum>
<enum name="sound_only_one">
<para>The sound played to a user when there is only one other
person is in the conference.</para>
</enum>
<enum name="sound_there_are">
<para>The sound played when announcing how many users there
are in a conference.</para>
</enum>
<enum name="sound_other_in_party">
<para>This file is used in conjunction with <literal>sound_there_are</literal>
when announcing how many users there are in the conference.
The sounds are stringed together like this.
<literal>"sound_there_are" ${number of participants} "sound_other_in_party"</literal>
</para>
</enum>
<enum name="sound_place_into_conference">
<para>The sound played when someone is placed into the conference
after waiting for a marked user.</para>
</enum>
<enum name="sound_wait_for_leader">
<para>The sound played when a user is placed into a conference that
can not start until a marked user enters.</para>
</enum>
<enum name="sound_leader_has_left">
<para>The sound played when the last marked user leaves the conference.</para>
</enum>
<enum name="sound_get_pin">
<para>The sound played when prompting for a conference pin number.</para>
</enum>
<enum name="sound_invalid_pin">
<para>The sound played when an invalid pin is entered too many times.</para>
</enum>
<enum name="sound_locked">
<para>The sound played to a user trying to join a locked conference.</para>
</enum>
<enum name="sound_locked_now">
<para>The sound played to an admin after toggling the conference to locked mode.</para>
</enum>
<enum name="sound_unlocked_now">
<para>The sound played to an admin after toggling the conference to unlocked mode.</para>
</enum>
<enum name="sound_error_menu">
<para>The sound played when an invalid menu option is entered.</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="template">
<synopsis>When using the CONFBRIDGE dialplan function, use a bridge profile as a template for creating a new temporary profile</synopsis>
</configOption>
</configObject>
<configObject name="menu">
<synopsis>A conference user menu</synopsis>
<description>
<para>Conference users, as defined by a <replaceable>conf_user</replaceable>,
can have a DTMF menu assigned to their profile when they enter the
<literal>ConfBridge</literal> application.</para>
</description>
<configOption name="type">
<synopsis>Define this configuration category as a menu</synopsis>
<description>
<para>The type parameter determines how a context in the
configuration file is interpreted.</para>
<enumlist>
<enum name="user">
<para>Configure the context as a <replaceable>user_profile</replaceable></para>
</enum>
<enum name="bridge">
<para>Configure the context as a <replaceable>bridge_profile</replaceable></para>
</enum>
<enum name="menu">
<para>Configure the context as a <replaceable>menu</replaceable></para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="template">
<synopsis>When using the CONFBRIDGE dialplan function, use a menu profile as a template for creating a new temporary profile</synopsis>
</configOption>
<configOption name="^[0-9A-D*#]+$">
<synopsis>DTMF sequences to assign various confbridge actions to</synopsis>
<description>
<para>The ConfBridge application also has the ability to apply custom DTMF menus to
each channel using the application. Like the User and Bridge profiles a menu
is passed in to ConfBridge as an argument in the dialplan.</para>
<para>Below is a list of menu actions that can be assigned to a DTMF sequence.</para>
<note>
<para>
To have the first DTMF digit in a sequence be the '#' character, you need to
escape it. If it is not escaped then normal config file processing will
think it is a directive like #include. For example: The mute setting is
toggled when <literal>#1</literal> is pressed.</para>
<para>
<literal>\#1=toggle_mute</literal>
</para>
</note>
<note>
<para>
A single DTMF sequence can have multiple actions associated with it. This is
accomplished by stringing the actions together and using a <literal>,</literal> as the
delimiter. Example: Both listening and talking volume is reset when <literal>5</literal> is
pressed. <literal>5=reset_talking_volume, reset_listening_volume</literal>
</para>
</note>
<enumlist>
<enum name="playback(filename&amp;filename2&amp;...)">
<para>
<literal>playback</literal> will play back an audio file to a channel
and then immediately return to the conference.
This file can not be interupted by DTMF.
Multiple files can be chained together using the
<literal>&amp;</literal> character.</para>
</enum>
<enum name="playback_and_continue(filename&amp;filename2&amp;...)">
<para>
<literal>playback_and_continue</literal> will
play back a prompt while continuing to
collect the dtmf sequence. This is useful
when using a menu prompt that describes all
the menu options. Note however that any DTMF
during this action will terminate the prompts
playback. Prompt files can be chained together
using the <literal>&amp;</literal> character as a delimiter.</para>
</enum>
<enum name="toggle_mute">
<para>
Toggle turning on and off mute. Mute will make the user silent
to everyone else, but the user will still be able to listen in.
</para>
</enum>
<enum name="no_op">
<para>
This action does nothing (No Operation). Its only real purpose exists for
being able to reserve a sequence in the config as a menu exit sequence.</para>
</enum>
<enum name="decrease_listening_volume">
<para>
Decreases the channel's listening volume.</para>
</enum>
<enum name="increase_listening_volume">
<para>
Increases the channel's listening volume.</para>
</enum>
<enum name="reset_listening_volume">
<para>
Reset channel's listening volume to default level.</para>
</enum>
<enum name="decrease_talking_volume">
<para>
Decreases the channel's talking volume.</para>
</enum>
<enum name="increase_talking_volume">
<para>
Increases the channel's talking volume.</para>
</enum>
<enum name="reset_talking_volume">
<para>
Reset channel's talking volume to default level.</para>
</enum>
<enum name="dialplan_exec(context,exten,priority)">
<para>
The <literal>dialplan_exec</literal> action allows a user
to escape from the conference and execute
commands in the dialplan. Once the dialplan
exits the user will be put back into the
conference. The possibilities are endless!</para>
</enum>
<enum name="leave_conference">
<para>
This action allows a user to exit the conference and continue
execution in the dialplan.</para>
</enum>
<enum name="admin_kick_last">
<para>
This action allows an Admin to kick the last participant from the
conference. This action will only work for admins which allows
a single menu to be used for both users and admins.</para>
</enum>
<enum name="admin_toggle_conference_lock">
<para>
This action allows an Admin to toggle locking and
unlocking the conference. Non admins can not use
this action even if it is in their menu.</para>
</enum>
<enum name="set_as_single_video_src">
<para>
This action allows any user to set themselves as the
single video source distributed to all participants.
This will make the video feed stick to them regardless
of what the <literal>video_mode</literal> is set to.</para>
</enum>
<enum name="release_as_single_video_src">
<para>
This action allows a user to release themselves as
the video source. If <literal>video_mode</literal> is not set to <literal>none</literal>
this action will result in the conference returning to
whatever video mode the bridge profile is using.</para>
<para>Note that this action will have no effect if the user
is not currently the video source. Also, the user is
not guaranteed by using this action that they will not
become the video source again. The bridge will return
to whatever operation the <literal>video_mode</literal> option is set to
upon release of the video src.</para>
</enum>
<enum name="admin_toggle_mute_participants">
<para>
This action allows an administrator to toggle the mute
state for all non-admins within a conference. All
admin users are unaffected by this option. Note that all
users, regardless of their admin status, are notified
that the conference is muted.</para>
</enum>
<enum name="participant_count">
<para>
This action plays back the number of participants currently
in a conference</para>
</enum>
</enumlist>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<managerEvent language="en_US" name="ConfbridgeStart">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a conference starts.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeEnd</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeEnd">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a conference ends.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeStart</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeJoin">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel joins a Confbridge conference.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
<parameter name="Muted">
<para>The joining mute status.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeLeave</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeLeave">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel leaves a Confbridge conference.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeJoin</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeRecord">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a conference starts recording.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeStopRecord</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeStopRecord">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a conference that was recording stops recording.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeRecord</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeMute">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a Confbridge participant mutes.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeUnmute</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeUnmute">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a confbridge participant unmutes.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">ConfbridgeMute</ref>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeTalking">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a confbridge participant begins or ends talking.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
</parameter>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="TalkingStatus">
<enumlist>
<enum name="on"/>
<enum name="off"/>
</enumlist>
</parameter>
<parameter name="Admin">
<para>Identifies this user as an admin user.</para>
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="application">ConfBridge</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<application name="BackgroundDetect" language="en_US">
<synopsis>
Background a file with talk detect.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="sil">
<para>If not specified, defaults to <literal>1000</literal>.</para>
</parameter>
<parameter name="min">
<para>If not specified, defaults to <literal>100</literal>.</para>
</parameter>
<parameter name="max">
<para>If not specified, defaults to <literal>infinity</literal>.</para>
</parameter>
<parameter name="analysistime">
<para>If not specified, defaults to <literal>infinity</literal>.</para>
</parameter>
</syntax>
<description>
<para>Plays back <replaceable>filename</replaceable>, waiting for interruption from a given digit (the digit
must start the beginning of a valid extension, or it will be ignored). During
the playback of the file, audio is monitored in the receive direction, and if
a period of non-silence which is greater than <replaceable>min</replaceable> ms yet less than
<replaceable>max</replaceable> ms is followed by silence for at least <replaceable>sil</replaceable> ms,
which occurs during the first <replaceable>analysistime</replaceable> ms, then the audio playback is
aborted and processing jumps to the <replaceable>talk</replaceable> extension, if available.</para>
</description>
</application>
<application name="CELGenUserEvent" language="en_US">
<synopsis>
Generates a CEL User Defined Event.
</synopsis>
<syntax>
<parameter name="event-name" required="true">
<argument name="event-name" required="true">
</argument>
<argument name="extra" required="false">
<para>Extra text to be included with the event.</para>
</argument>
</parameter>
</syntax>
<description>
<para>A CEL event will be immediately generated by this channel, with the supplied name for a type.</para>
</description>
</application>
<application name="Dictate" language="en_US">
<synopsis>
Virtual Dictation Machine.
</synopsis>
<syntax>
<parameter name="base_dir" />
<parameter name="filename" />
</syntax>
<description>
<para>Start dictation machine using optional <replaceable>base_dir</replaceable> for files.</para>
</description>
</application>
<application name="NBScat" language="en_US">
<synopsis>
Play an NBS local stream.
</synopsis>
<syntax />
<description>
<para>Executes nbscat to listen to the local NBS stream.
User can exit by pressing any key.</para>
</description>
</application>
<application name="SendFAX" language="en_US" module="app_fax">
<synopsis>
Send a Fax
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>Filename of TIFF file to fax</para>
</parameter>
<parameter name="a" required="false">
<para>Makes the application behave as the answering machine</para>
<para>(Default behavior is as calling machine)</para>
</parameter>
</syntax>
<description>
<para>Send a given TIFF file to the channel as a FAX.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="LOCALSTATIONID">
<para>To identify itself to the remote end</para>
</variable>
<variable name="LOCALHEADERINFO">
<para>To generate a header line on each page</para>
</variable>
<variable name="FAXSTATUS">
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
<variable name="FAXERROR">
<para>Cause of failure</para>
</variable>
<variable name="REMOTESTATIONID">
<para>The CSID of the remote side</para>
</variable>
<variable name="FAXPAGES">
<para>Number of pages sent</para>
</variable>
<variable name="FAXBITRATE">
<para>Transmission rate</para>
</variable>
<variable name="FAXRESOLUTION">
<para>Resolution of sent fax</para>
</variable>
</variablelist>
</description>
</application>
<application name="ReceiveFAX" language="en_US" module="app_fax">
<synopsis>
Receive a Fax
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>Filename of TIFF file save incoming fax</para>
</parameter>
<parameter name="c" required="false">
<para>Makes the application behave as the calling machine</para>
<para>(Default behavior is as answering machine)</para>
</parameter>
</syntax>
<description>
<para>Receives a FAX from the channel into the given filename
overwriting the file if it already exists.</para>
<para>File created will be in TIFF format.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="LOCALSTATIONID">
<para>To identify itself to the remote end</para>
</variable>
<variable name="LOCALHEADERINFO">
<para>To generate a header line on each page</para>
</variable>
<variable name="FAXSTATUS">
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
<variable name="FAXERROR">
<para>Cause of failure</para>
</variable>
<variable name="REMOTESTATIONID">
<para>The CSID of the remote side</para>
</variable>
<variable name="FAXPAGES">
<para>Number of pages sent</para>
</variable>
<variable name="FAXBITRATE">
<para>Transmission rate</para>
</variable>
<variable name="FAXRESOLUTION">
<para>Resolution of sent fax</para>
</variable>
</variablelist>
</description>
</application>
<application name="While" language="en_US">
<synopsis>
Start a while loop.
</synopsis>
<syntax>
<parameter name="expr" required="true" />
</syntax>
<description>
<para>Start a While Loop. Execution will return to this point when
<literal>EndWhile()</literal> is called until expr is no longer true.</para>
</description>
<see-also>
<ref type="application">EndWhile</ref>
<ref type="application">ExitWhile</ref>
<ref type="application">ContinueWhile</ref>
</see-also>
</application>
<application name="EndWhile" language="en_US">
<synopsis>
End a while loop.
</synopsis>
<syntax />
<description>
<para>Return to the previous called <literal>While()</literal>.</para>
</description>
<see-also>
<ref type="application">While</ref>
<ref type="application">ExitWhile</ref>
<ref type="application">ContinueWhile</ref>
</see-also>
</application>
<application name="ExitWhile" language="en_US">
<synopsis>
End a While loop.
</synopsis>
<syntax />
<description>
<para>Exits a <literal>While()</literal> loop, whether or not the conditional has been satisfied.</para>
</description>
<see-also>
<ref type="application">While</ref>
<ref type="application">EndWhile</ref>
<ref type="application">ContinueWhile</ref>
</see-also>
</application>
<application name="ContinueWhile" language="en_US">
<synopsis>
Restart a While loop.
</synopsis>
<syntax />
<description>
<para>Returns to the top of the while loop and re-evaluates the conditional.</para>
</description>
<see-also>
<ref type="application">While</ref>
<ref type="application">EndWhile</ref>
<ref type="application">ExitWhile</ref>
</see-also>
</application>
<application name="ReadExten" language="en_US">
<synopsis>
Read an extension into a variable.
</synopsis>
<syntax>
<parameter name="variable" required="true" />
<parameter name="filename">
<para>File to play before reading digits or tone with option <literal>i</literal></para>
</parameter>
<parameter name="context">
<para>Context in which to match extensions.</para>
</parameter>
<parameter name="option">
<optionlist>
<option name="s">
<para>Return immediately if the channel is not answered.</para>
</option>
<option name="i">
<para>Play <replaceable>filename</replaceable> as an indication tone from your
<filename>indications.conf</filename> or a directly specified list of
frequencies and durations.</para>
</option>
<option name="n">
<para>Read digits even if the channel is not answered.</para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>An integer number of seconds to wait for a digit response. If
greater than <literal>0</literal>, that value will override the default timeout.</para>
</parameter>
</syntax>
<description>
<para>Reads a <literal>#</literal> terminated string of digits from the user into the given variable.</para>
<para>Will set READEXTENSTATUS on exit with one of the following statuses:</para>
<variablelist>
<variable name="READEXTENSTATUS">
<value name="OK">
A valid extension exists in ${variable}.
</value>
<value name="TIMEOUT">
No extension was entered in the specified time. Also sets ${variable} to "t".
</value>
<value name="INVALID">
An invalid extension, ${INVALID_EXTEN}, was entered. Also sets ${variable} to "i".
</value>
<value name="SKIP">
Line was not up and the option 's' was specified.
</value>
<value name="ERROR">
Invalid arguments were passed.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="Festival" language="en_US">
<synopsis>
Say text to the user.
</synopsis>
<syntax>
<parameter name="text" required="true" />
<parameter name="intkeys" />
</syntax>
<description>
<para>Connect to Festival, send the argument, get back the waveform, play it to the user,
allowing any given interrupt keys to immediately terminate and return the value, or
<literal>any</literal> to allow any number back (useful in dialplan).</para>
</description>
</application>
<application name="FollowMe" language="en_US">
<synopsis>
Find-Me/Follow-Me application.
</synopsis>
<syntax>
<parameter name="followmeid" required="true" />
<parameter name="options">
<optionlist>
<option name="a">
<para>Record the caller's name so it can be announced to the
callee on each step.</para>
</option>
<option name="B" argsep="^">
<para>Before initiating the outgoing call(s), Gosub to the specified
location using the current channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="b" argsep="^">
<para>Before initiating an outgoing call, Gosub to the specified
location using the newly created channel. The Gosub will be
executed for each destination channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="d">
<para>Disable the 'Please hold while we try to connect your call' announcement.</para>
</option>
<option name="I">
<para>Asterisk will ignore any connected line update requests
it may receive on this dial attempt.</para>
</option>
<option name="l">
<para>Disable local call optimization so that applications with
audio hooks between the local bridge don't get dropped when the
calls get joined directly.</para>
</option>
<option name="N">
<para>Don't answer the incoming call until we're ready to
connect the caller or give up.</para>
<note>
<para>This option is ignored if the call is already answered.</para>
</note>
<note>
<para>If the call is not already answered, the 'a' and 's'
options are ignored while the 'd' option is implicitly enabled.</para>
</note>
</option>
<option name="n">
<para>Playback the unreachable status message if we've run out
of steps or the callee has elected not to be reachable.</para>
</option>
<option name="s">
<para>Playback the incoming status message prior to starting
the follow-me step(s)</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application performs Find-Me/Follow-Me functionality for the caller
as defined in the profile matching the <replaceable>followmeid</replaceable> parameter in
<filename>followme.conf</filename>. If the specified <replaceable>followmeid</replaceable>
profile doesn't exist in <filename>followme.conf</filename>, execution will be returned
to the dialplan and call execution will continue at the next priority.</para>
<para>Returns -1 on hangup.</para>
</description>
</application>
<application name="Morsecode" language="en_US">
<synopsis>
Plays morse code.
</synopsis>
<syntax>
<parameter name="string" required="true">
<para>String to playback as morse code to channel</para>
</parameter>
</syntax>
<description>
<para>Plays the Morse code equivalent of the passed string.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer() or Progress().</para>
<para>This application uses the following variables:</para>
<variablelist>
<variable name="MORSEDITLEN">
<para>Use this value in (ms) for length of dit</para>
</variable>
<variable name="MORSETONE">
<para>The pitch of the tone in (Hz), default is 800</para>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">SayAlpha</ref>
<ref type="application">SayPhonetic</ref>
</see-also>
</application>
<application name="SayUnixTime" language="en_US">
<synopsis>
Says a specified time in a custom format.
</synopsis>
<syntax>
<parameter name="unixtime" required="false">
<para>time, in seconds since Jan 1, 1970. May be negative. Defaults to now.</para>
</parameter>
<parameter name="timezone" required="false" >
<para>timezone, see <directory>/usr/share/zoneinfo</directory> for a list. Defaults to machine default.</para>
</parameter>
<parameter name="format" required="false" >
<para>a format the time is to be said in. See <filename>voicemail.conf</filename>.
Defaults to <literal>ABdY "digits/at" IMp</literal>
</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="j">
<para>Allow the calling user to dial digits to jump to that extension.
This option is automatically enabled if
<variable>SAY_DTMF_INTERRUPT</variable> is present on the channel and
set to 'true' (case insensitive)</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Uses some of the sound files stored in <directory>/var/lib/asterisk/sounds</directory> to construct a phrase
saying the specified date and/or time in the specified format. </para>
</description>
<see-also>
<ref type="function">STRFTIME</ref>
<ref type="function">STRPTIME</ref>
<ref type="function">IFTIME</ref>
</see-also>
</application>
<application name="DateTime" language="en_US">
<synopsis>
Says a specified time in a custom format.
</synopsis>
<syntax>
<parameter name="unixtime">
<para>time, in seconds since Jan 1, 1970. May be negative. Defaults to now.</para>
</parameter>
<parameter name="timezone">
<para>timezone, see <filename>/usr/share/zoneinfo</filename> for a list. Defaults to machine default.</para>
</parameter>
<parameter name="format">
<para>a format the time is to be said in. See <filename>voicemail.conf</filename>.
Defaults to <literal>ABdY "digits/at" IMp</literal>
</para>
</parameter>
</syntax>
<description>
<para>Say the date and time in a specified format.</para>
</description>
</application>
<application name="DISA" language="en_US">
<synopsis>
Direct Inward System Access.
</synopsis>
<syntax>
<parameter name="passcode|filename" required="true">
<para>If you need to present a DISA dialtone without entering a password,
simply set <replaceable>passcode</replaceable> to <literal>no-password</literal>
</para>
<para>You may specified a <replaceable>filename</replaceable> instead of a
<replaceable>passcode</replaceable>, this filename must contain individual passcodes</para>
</parameter>
<parameter name="context">
<para>Specifies the dialplan context in which the user-entered extension
will be matched. If no context is specified, the DISA application defaults
to the <literal>disa</literal> context. Presumably a normal system will have a special
context set up for DISA use with some or a lot of restrictions.</para>
</parameter>
<parameter name="cid">
<para>Specifies a new (different) callerid to be used for this call.</para>
</parameter>
<parameter name="mailbox" argsep="@">
<para>Will cause a stutter-dialtone (indication <emphasis>dialrecall</emphasis>)
to be used, if the specified mailbox contains any new messages.</para>
<argument name="mailbox" required="true" />
<argument name="context" required="false" />
</parameter>
<parameter name="options">
<optionlist>
<option name="n">
<para>The DISA application will not answer initially.</para>
</option>
<option name="p">
<para>The extension entered will be considered complete when a <literal>#</literal>
is entered.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The DISA, Direct Inward System Access, application allows someone from
outside the telephone switch (PBX) to obtain an <emphasis>internal</emphasis> system
dialtone and to place calls from it as if they were placing a call from
within the switch.
DISA plays a dialtone. The user enters their numeric passcode, followed by
the pound sign <literal>#</literal>. If the passcode is correct, the user is then given
system dialtone within <replaceable>context</replaceable> on which a call may be placed.
If the user enters an invalid extension and extension <literal>i</literal> exists in the specified
<replaceable>context</replaceable>, it will be used.
</para>
<para>Be aware that using this may compromise the security of your PBX.</para>
<para>The arguments to this application (in <filename>extensions.conf</filename>) allow either
specification of a single global <replaceable>passcode</replaceable> (that everyone uses), or
individual passcodes contained in a file (<replaceable>filename</replaceable>).</para>
<para>The file that contains the passcodes (if used) allows a complete
specification of all of the same arguments available on the command
line, with the sole exception of the options. The file may contain blank
lines, or comments starting with <literal>#</literal> or <literal>;</literal>.</para>
</description>
<see-also>
<ref type="application">Authenticate</ref>
<ref type="application">VMAuthenticate</ref>
</see-also>
</application>
<application name="ChannelRedirect" language="en_US">
<synopsis>
Redirects given channel to a dialplan target
</synopsis>
<syntax>
<parameter name="channel" required="true" />
<parameter name="context" required="false" />
<parameter name="extension" required="false" />
<parameter name="priority" required="true" />
</syntax>
<description>
<para>Sends the specified channel to the specified extension priority</para>
<para>This application sets the following channel variables upon completion</para>
<variablelist>
<variable name="CHANNELREDIRECT_STATUS">
<value name="NOCHANNEL" />
<value name="SUCCESS" />
<para>Are set to the result of the redirection</para>
</variable>
</variablelist>
</description>
</application>
<application name="MinivmRecord" language="en_US">
<synopsis>
Receive Mini-Voicemail and forward via e-mail.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="username" required="true">
<para>Voicemail username</para>
</argument>
<argument name="domain" required="true">
<para>Voicemail domain</para>
</argument>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="0">
<para>Jump to the <literal>o</literal> extension in the current dialplan context.</para>
</option>
<option name="*">
<para>Jump to the <literal>a</literal> extension in the current dialplan context.</para>
</option>
<option name="g">
<argument name="gain">
<para>Amount of gain to use</para>
</argument>
<para>Use the specified amount of gain when recording the voicemail message.
The units are whole-number decibels (dB).</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in <filename>minivm.conf</filename></para>
<para>MiniVM records audio file in configured format and forwards message to e-mail and pager.</para>
<para>If there's no user account for that address, a temporary account will be used with default options.</para>
<para>The recorded file name and path will be stored in <variable>MVM_FILENAME</variable> and the duration
of the message will be stored in <variable>MVM_DURATION</variable>
</para>
<note>
<para>If the caller hangs up after the recording, the only way to send the message and clean up is to
execute in the <literal>h</literal> extension. The application will exit if any of the following DTMF digits
are received and the requested extension exist in the current context.</para>
</note>
<variablelist>
<variable name="MVM_RECORD_STATUS">
<para>This is the status of the record operation</para>
<value name="SUCCESS" />
<value name="USEREXIT" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
</application>
<application name="MinivmGreet" language="en_US">
<synopsis>
Play Mini-Voicemail prompts.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="username" required="true">
<para>Voicemail username</para>
</argument>
<argument name="domain" required="true">
<para>Voicemail domain</para>
</argument>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="b">
<para>Play the <literal>busy</literal> greeting to the calling party.</para>
</option>
<option name="s">
<para>Skip the playback of instructions for leaving a message to the calling party.</para>
</option>
<option name="u">
<para>Play the <literal>unavailable</literal> greeting.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in minivm.conf.</para>
<para>MinivmGreet() plays default prompts or user specific prompts for an account.</para>
<para>Busy and unavailable messages can be choosen, but will be overridden if a temporary
message exists for the account.</para>
<variablelist>
<variable name="MVM_GREET_STATUS">
<para>This is the status of the greeting playback.</para>
<value name="SUCCESS" />
<value name="USEREXIT" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
</application>
<application name="MinivmNotify" language="en_US">
<synopsis>
Notify voicemail owner about new messages.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="username" required="true">
<para>Voicemail username</para>
</argument>
<argument name="domain" required="true">
<para>Voicemail domain</para>
</argument>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="template">
<para>E-mail template to use for voicemail notification</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in minivm.conf.</para>
<para>MiniVMnotify forwards messages about new voicemail to e-mail and pager. If there's no user
account for that address, a temporary account will be used with default options (set in
<filename>minivm.conf</filename>).</para>
<para>If the channel variable <variable>MVM_COUNTER</variable> is set, this will be used in the message
file name and available in the template for the message.</para>
<para>If no template is given, the default email template will be used to send email and default pager
template to send paging message (if the user account is configured with a paging address.</para>
<variablelist>
<variable name="MVM_NOTIFY_STATUS">
<para>This is the status of the notification attempt</para>
<value name="SUCCESS" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
</application>
<application name="MinivmDelete" language="en_US">
<synopsis>
Delete Mini-Voicemail voicemail messages.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>File to delete</para>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in <filename>minivm.conf</filename>.</para>
<para>It deletes voicemail file set in MVM_FILENAME or given filename.</para>
<variablelist>
<variable name="MVM_DELETE_STATUS">
<para>This is the status of the delete operation.</para>
<value name="SUCCESS" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
</application>
<application name="MinivmAccMess" language="en_US">
<synopsis>
Record account specific messages.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="username" required="true">
<para>Voicemail username</para>
</argument>
<argument name="domain" required="true">
<para>Voicemail domain</para>
</argument>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="u">
<para>Record the <literal>unavailable</literal> greeting.</para>
</option>
<option name="b">
<para>Record the <literal>busy</literal> greeting.</para>
</option>
<option name="t">
<para>Record the temporary greeting.</para>
</option>
<option name="n">
<para>Account name.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in <filename>minivm.conf</filename>.</para>
<para>Use this application to record account specific audio/video messages for busy, unavailable
and temporary messages.</para>
<para>Account specific directories will be created if they do not exist.</para>
<variablelist>
<variable name="MVM_ACCMESS_STATUS">
<para>This is the result of the attempt to record the specified greeting.</para>
<para>
<literal>FAILED</literal> is set if the file can't be created.</para>
<value name="SUCCESS" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
</application>
<application name="MinivmMWI" language="en_US">
<synopsis>
Send Message Waiting Notification to subscriber(s) of mailbox.
</synopsis>
<syntax>
<parameter name="mailbox" required="true" argsep="@">
<argument name="username" required="true">
<para>Voicemail username</para>
</argument>
<argument name="domain" required="true">
<para>Voicemail domain</para>
</argument>
</parameter>
<parameter name="urgent" required="true">
<para>Number of urgent messages in mailbox.</para>
</parameter>
<parameter name="new" required="true">
<para>Number of new messages in mailbox.</para>
</parameter>
<parameter name="old" required="true">
<para>Number of old messages in mailbox.</para>
</parameter>
</syntax>
<description>
<para>This application is part of the Mini-Voicemail system, configured in <filename>minivm.conf</filename>.</para>
<para>MinivmMWI is used to send message waiting indication to any devices whose channels have
subscribed to the mailbox passed in the first parameter.</para>
</description>
</application>
<function name="MINIVMCOUNTER" language="en_US">
<synopsis>
Reads or sets counters for MiniVoicemail message.
</synopsis>
<syntax argsep=":">
<parameter name="account" required="true">
<para>If account is given and it exists, the counter is specific for the account.</para>
<para>If account is a domain and the domain directory exists, counters are specific for a domain.</para>
</parameter>
<parameter name="name" required="true">
<para>The name of the counter is a string, up to 10 characters.</para>
</parameter>
<parameter name="operand">
<para>The counters never goes below zero. Valid operands for changing the value of a counter when assigning a value are:</para>
<enumlist>
<enum name="i">
<para>Increment by value.</para>
</enum>
<enum name="d">
<para>Decrement by value.</para>
</enum>
<enum name="s">
<para>Set to value.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>The operation is atomic and the counter is locked while changing the value. The counters are stored as text files in the minivm account directories. It might be better to use realtime functions if you are using a database to operate your Asterisk.</para>
</description>
<see-also>
<ref type="application">MinivmRecord</ref>
<ref type="application">MinivmGreet</ref>
<ref type="application">MinivmNotify</ref>
<ref type="application">MinivmDelete</ref>
<ref type="application">MinivmAccMess</ref>
<ref type="application">MinivmMWI</ref>
<ref type="function">MINIVMACCOUNT</ref>
</see-also>
</function>
<function name="MINIVMACCOUNT" language="en_US">
<synopsis>
Gets MiniVoicemail account information.
</synopsis>
<syntax argsep=":">
<parameter name="account" required="true" />
<parameter name="item" required="true">
<para>Valid items are:</para>
<enumlist>
<enum name="path">
<para>Path to account mailbox (if account exists, otherwise temporary mailbox).</para>
</enum>
<enum name="hasaccount">
<para>1 is static Minivm account exists, 0 otherwise.</para>
</enum>
<enum name="fullname">
<para>Full name of account owner.</para>
</enum>
<enum name="email">
<para>Email address used for account.</para>
</enum>
<enum name="etemplate">
<para>Email template for account (default template if none is configured).</para>
</enum>
<enum name="ptemplate">
<para>Pager template for account (default template if none is configured).</para>
</enum>
<enum name="accountcode">
<para>Account code for the voicemail account.</para>
</enum>
<enum name="pincode">
<para>Pin code for voicemail account.</para>
</enum>
<enum name="timezone">
<para>Time zone for voicemail account.</para>
</enum>
<enum name="language">
<para>Language for voicemail account.</para>
</enum>
<enum name="&lt;channel variable name&gt;">
<para>Channel variable value (set in configuration for account).</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para />
</description>
<see-also>
<ref type="application">MinivmRecord</ref>
<ref type="application">MinivmGreet</ref>
<ref type="application">MinivmNotify</ref>
<ref type="application">MinivmDelete</ref>
<ref type="application">MinivmAccMess</ref>
<ref type="application">MinivmMWI</ref>
<ref type="function">MINIVMCOUNTER</ref>
</see-also>
</function>
<managerEvent language="en_US" name="MiniVoiceMail">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a notification is sent out by a MiniVoiceMail application</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Action">
<para>What action was taken. Currently, this will always be <literal>SentNotification</literal></para>
</parameter>
<parameter name="Mailbox">
<para>The mailbox that the notification was about, specified as <literal>mailbox</literal>@<literal>context</literal></para>
</parameter>
<parameter name="Counter">
<para>A message counter derived from the <literal>MVM_COUNTER</literal> channel variable.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<application name="JACK" language="en_US">
<synopsis>
Jack Audio Connection Kit
</synopsis>
<syntax>
<parameter name="options" required="false">
<optionlist>
<option name="s">
<argument name="name" required="true">
<para>Connect to the specified jack server name</para>
</argument>
</option>
<option name="i">
<argument name="name" required="true">
<para>Connect the output port that gets created to the specified jack input port</para>
</argument>
</option>
<option name="o">
<argument name="name" required="true">
<para>Connect the input port that gets created to the specified jack output port</para>
</argument>
</option>
<option name="c">
<argument name="name" required="true">
<para>By default, Asterisk will use the channel name for the jack client name.</para>
<para>Use this option to specify a custom client name.</para>
</argument>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>When executing this application, two jack ports will be created;
one input and one output. Other applications can be hooked up to
these ports to access audio coming from, or being send to the channel.</para>
</description>
</application>
<application name="Milliwatt" language="en_US">
<synopsis>
Generate a Constant 1004Hz tone at 0dbm (mu-law).
</synopsis>
<syntax>
<parameter name="options">
<optionlist>
<option name="o">
<para>Generate the tone at 1000Hz like previous version.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Previous versions of this application generated the tone at 1000Hz. If for
some reason you would prefer that behavior, supply the <literal>o</literal> option to get the
old behavior.</para>
</description>
</application>
<application name="ChanSpy" language="en_US">
<synopsis>
Listen to a channel, and optionally whisper into it.
</synopsis>
<syntax>
<parameter name="chanprefix" />
<parameter name="options">
<optionlist>
<option name="b">
<para>Only spy on channels involved in a bridged call.</para>
</option>
<option name="B">
<para>Instead of whispering on a single channel barge in on both
channels involved in the call.</para>
</option>
<option name="c">
<argument name="digit" required="true">
<para>Specify a DTMF digit that can be used to spy on the next available channel.</para>
</argument>
</option>
<option name="d">
<para>Override the typical numeric DTMF functionality and instead
use DTMF to switch between spy modes.</para>
<enumlist>
<enum name="4">
<para>spy mode</para>
</enum>
<enum name="5">
<para>whisper mode</para>
</enum>
<enum name="6">
<para>barge mode</para>
</enum>
</enumlist>
</option>
<option name="e">
<argument name="ext" required="true" />
<para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
only monitor extensions whose name is in the <replaceable>ext</replaceable> : delimited
list.</para>
</option>
<option name="E">
<para>Exit when the spied-on channel hangs up.</para>
</option>
<option name="g">
<argument name="grp" required="true">
<para>Only spy on channels in which one or more of the groups
listed in <replaceable>grp</replaceable> matches one or more groups from the
<variable>SPYGROUP</variable> variable set on the channel to be spied upon.</para>
</argument>
<note>
<para>both <replaceable>grp</replaceable> and <variable>SPYGROUP</variable> can contain
either a single group or a colon-delimited list of groups, such
as <literal>sales:support:accounting</literal>.</para>
</note>
</option>
<option name="n" argsep="@">
<para>Say the name of the person being spied on if that person has recorded
his/her name. If a context is specified, then that voicemail context will
be searched when retrieving the name, otherwise the <literal>default</literal> context
be used when searching for the name (i.e. if SIP/1000 is the channel being
spied on and no mailbox is specified, then <literal>1000</literal> will be used when searching
for the name).</para>
<argument name="mailbox" />
<argument name="context" />
</option>
<option name="o">
<para>Only listen to audio coming from this channel.</para>
</option>
<option name="q">
<para>Don't play a beep when beginning to spy on a channel, or speak the
selected channel name.</para>
</option>
<option name="r">
<para>Record the session to the monitor spool directory. An optional base for the filename
may be specified. The default is <literal>chanspy</literal>.</para>
<argument name="basename" />
</option>
<option name="s">
<para>Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.</para>
</option>
<option name="S">
<para>Stop when no more channels are left to spy on.</para>
</option>
<option name="u">
<para>The <literal>chanprefix</literal> parameter is a channel uniqueid
or fully specified channel name.</para>
</option>
<option name="v">
<argument name="value" />
<para>Adjust the initial volume in the range from <literal>-4</literal>
to <literal>4</literal>. A negative value refers to a quieter setting.</para>
</option>
<option name="w">
<para>Enable <literal>whisper</literal> mode, so the spying channel can talk to
the spied-on channel.</para>
</option>
<option name="W">
<para>Enable <literal>private whisper</literal> mode, so the spying channel can
talk to the spied-on channel but cannot listen to that channel.</para>
</option>
<option name="x">
<argument name="digit" required="true">
<para>Specify a DTMF digit that can be used to exit the application while actively
spying on a channel. If there is no channel being spied on, the DTMF digit will be
ignored.</para>
</argument>
</option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
numeric extension in the current context or the context
specified by the <variable>SPY_EXIT_CONTEXT</variable> channel variable. The
name of the last channel that was spied on will be stored
in the <variable>SPY_CHANNEL</variable> variable.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is used to listen to the audio from an Asterisk channel. This includes the audio
coming in and out of the channel being spied on. If the <literal>chanprefix</literal> parameter is specified,
only channels beginning with this string will be spied upon.</para>
<para>While spying, the following actions may be performed:</para>
<para> - Dialing <literal>#</literal> cycles the volume level.</para>
<para> - Dialing <literal>*</literal> will stop spying and look for another channel to spy on.</para>
<para> - Dialing a series of digits followed by <literal>#</literal> builds a channel name to append
to <literal>chanprefix</literal>. For example, executing ChanSpy(Agent) and then dialing the digits '1234#'
while spying will begin spying on the channel 'Agent/1234'. Note that this feature will be overridden
if the 'd' or 'u' options are used.</para>
<note>
<para>The <replaceable>X</replaceable> option supersedes the three features above in that if a valid
single digit extension exists in the correct context ChanSpy will exit to it.
This also disables choosing a channel based on <literal>chanprefix</literal> and a digit sequence.</para>
</note>
</description>
<see-also>
<ref type="application">ExtenSpy</ref>
<ref type="managerEvent">ChanSpyStart</ref>
<ref type="managerEvent">ChanSpyStop</ref>
</see-also>
</application>
<application name="ExtenSpy" language="en_US">
<synopsis>
Listen to a channel, and optionally whisper into it.
</synopsis>
<syntax>
<parameter name="exten" required="true" argsep="@">
<argument name="exten" required="true">
<para>Specify extension.</para>
</argument>
<argument name="context">
<para>Optionally specify a context, defaults to <literal>default</literal>.</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="b">
<para>Only spy on channels involved in a bridged call.</para>
</option>
<option name="B">
<para>Instead of whispering on a single channel barge in on both
channels involved in the call.</para>
</option>
<option name="c">
<argument name="digit" required="true">
<para>Specify a DTMF digit that can be used to spy on the next available channel.</para>
</argument>
</option>
<option name="d">
<para>Override the typical numeric DTMF functionality and instead
use DTMF to switch between spy modes.</para>
<enumlist>
<enum name="4">
<para>spy mode</para>
</enum>
<enum name="5">
<para>whisper mode</para>
</enum>
<enum name="6">
<para>barge mode</para>
</enum>
</enumlist>
</option>
<option name="e">
<argument name="ext" required="true" />
<para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
only monitor extensions whose name is in the <replaceable>ext</replaceable> : delimited
list.</para>
</option>
<option name="E">
<para>Exit when the spied-on channel hangs up.</para>
</option>
<option name="g">
<argument name="grp" required="true">
<para>Only spy on channels in which one or more of the groups
listed in <replaceable>grp</replaceable> matches one or more groups from the
<variable>SPYGROUP</variable> variable set on the channel to be spied upon.</para>
</argument>
<note>
<para>both <replaceable>grp</replaceable> and <variable>SPYGROUP</variable> can contain
either a single group or a colon-delimited list of groups, such
as <literal>sales:support:accounting</literal>.</para>
</note>
</option>
<option name="n" argsep="@">
<para>Say the name of the person being spied on if that person has recorded
his/her name. If a context is specified, then that voicemail context will
be searched when retrieving the name, otherwise the <literal>default</literal> context
be used when searching for the name (i.e. if SIP/1000 is the channel being
spied on and no mailbox is specified, then <literal>1000</literal> will be used when searching
for the name).</para>
<argument name="mailbox" />
<argument name="context" />
</option>
<option name="o">
<para>Only listen to audio coming from this channel.</para>
</option>
<option name="q">
<para>Don't play a beep when beginning to spy on a channel, or speak the
selected channel name.</para>
</option>
<option name="r">
<para>Record the session to the monitor spool directory. An optional base for the filename
may be specified. The default is <literal>chanspy</literal>.</para>
<argument name="basename" />
</option>
<option name="s">
<para>Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.</para>
</option>
<option name="S">
<para>Stop when there are no more extensions left to spy on.</para>
</option>
<option name="v">
<argument name="value" />
<para>Adjust the initial volume in the range from <literal>-4</literal>
to <literal>4</literal>. A negative value refers to a quieter setting.</para>
</option>
<option name="w">
<para>Enable <literal>whisper</literal> mode, so the spying channel can talk to
the spied-on channel.</para>
</option>
<option name="W">
<para>Enable <literal>private whisper</literal> mode, so the spying channel can
talk to the spied-on channel but cannot listen to that channel.</para>
</option>
<option name="x">
<argument name="digit" required="true">
<para>Specify a DTMF digit that can be used to exit the application while actively
spying on a channel. If there is no channel being spied on, the DTMF digit will be
ignored.</para>
</argument>
</option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
numeric extension in the current context or the context
specified by the <variable>SPY_EXIT_CONTEXT</variable> channel variable. The
name of the last channel that was spied on will be stored
in the <variable>SPY_CHANNEL</variable> variable.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is used to listen to the audio from an Asterisk channel. This includes
the audio coming in and out of the channel being spied on. Only channels created by outgoing calls for the
specified extension will be selected for spying. If the optional context is not supplied,
the current channel's context will be used.</para>
<para>While spying, the following actions may be performed:</para>
<para> - Dialing <literal>#</literal> cycles the volume level.</para>
<para> - Dialing <literal>*</literal> will stop spying and look for another channel to spy on.</para>
<note>
<para>The <replaceable>X</replaceable> option supersedes the three features above in that if a valid
single digit extension exists in the correct context ChanSpy will exit to it.
This also disables choosing a channel based on <literal>chanprefix</literal> and a digit sequence.</para>
</note>
</description>
<see-also>
<ref type="application">ChanSpy</ref>
<ref type="managerEvent">ChanSpyStart</ref>
<ref type="managerEvent">ChanSpyStop</ref>
</see-also>
</application>
<application name="DAHDIScan" language="en_US">
<synopsis>
Scan DAHDI channels to monitor calls.
</synopsis>
<syntax>
<parameter name="group">
<para>Limit scanning to a channel <replaceable>group</replaceable> by setting this option.</para>
</parameter>
</syntax>
<description>
<para>Allows a call center manager to monitor DAHDI channels in a
convenient way. Use <literal>#</literal> to select the next channel and use <literal>*</literal> to exit.</para>
</description>
<see-also>
<ref type="managerEvent">ChanSpyStart</ref>
<ref type="managerEvent">ChanSpyStop</ref>
</see-also>
</application>
<application name="PrivacyManager" language="en_US">
<synopsis>
Require phone number to be entered, if no CallerID sent
</synopsis>
<syntax>
<parameter name="maxretries">
<para>Total tries caller is allowed to input a callerid. Defaults to <literal>3</literal>.</para>
</parameter>
<parameter name="minlength">
<para>Minimum allowable digits in the input callerid number. Defaults to <literal>10</literal>.</para>
</parameter>
<parameter name="options">
<para>Position reserved for options.</para>
</parameter>
<parameter name="context">
<para>Context to check the given callerid against patterns.</para>
</parameter>
</syntax>
<description>
<para>If no Caller*ID is sent, PrivacyManager answers the channel and asks
the caller to enter their phone number. The caller is given
<replaceable>maxretries</replaceable> attempts to do so. The application does
<emphasis>nothing</emphasis> if Caller*ID was received on the channel.</para>
<para>The application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PRIVACYMGRSTATUS">
<para>The status of the privacy manager's attempt to collect a phone number from the user.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">Zapateller</ref>
</see-also>
</application>
<application name="SendImage" language="en_US">
<synopsis>
Sends an image file.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>Path of the filename (image) to send.</para>
</parameter>
</syntax>
<description>
<para>Send an image file on a channel supporting it.</para>
<para>Result of transmission will be stored in <variable>SENDIMAGESTATUS</variable></para>
<variablelist>
<variable name="SENDIMAGESTATUS">
<value name="SUCCESS">
Transmission succeeded.
</value>
<value name="FAILURE">
Transmission failed.
</value>
<value name="UNSUPPORTED">
Image transmission not supported by channel.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">SendText</ref>
<ref type="application">SendURL</ref>
</see-also>
</application>
<application name="MeetMe" language="en_US">
<synopsis>
MeetMe conference bridge.
</synopsis>
<syntax>
<parameter name="confno">
<para>The conference number</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Set admin mode.</para>
</option>
<option name="A">
<para>Set marked mode.</para>
</option>
<option name="b">
<para>Run AGI script specified in <variable>MEETME_AGI_BACKGROUND</variable>
Default: <literal>conf-background.agi</literal>.</para>
<note>
<para>This does not work with non-DAHDI channels in the same
conference).</para>
</note>
</option>
<option name="c">
<para>Announce user(s) count on joining a conference.</para>
</option>
<option name="C">
<para>Continue in dialplan when kicked out of conference.</para>
</option>
<option name="d">
<para>Dynamically add conference.</para>
</option>
<option name="D">
<para>Dynamically add conference, prompting for a PIN.</para>
</option>
<option name="e">
<para>Select an empty conference.</para>
</option>
<option name="E">
<para>Select an empty pinless conference.</para>
</option>
<option name="F">
<para>Pass DTMF through the conference.</para>
</option>
<option name="G">
<argument name="x" required="true">
<para>The file to playback</para>
</argument>
<para>Play an intro announcement in conference.</para>
</option>
<option name="i">
<para>Announce user join/leave with review.</para>
</option>
<option name="I">
<para>Announce user join/leave without review.</para>
</option>
<option name="k">
<para>Close the conference if there's only one active participant left at exit.</para>
</option>
<option name="l">
<para>Set listen only mode (Listen only, no talking).</para>
</option>
<option name="m">
<para>Set initially muted.</para>
</option>
<option name="M" hasparams="optional">
<para>Enable music on hold when the conference has a single caller. Optionally,
specify a musiconhold class to use. If one is not provided, it will use the
channel's currently set music class, or <literal>default</literal>.</para>
<argument name="class" required="true" />
</option>
<option name="n">
<para>Disable the denoiser. By default, if <literal>func_speex</literal> is loaded, Asterisk
will apply a denoiser to channels in the MeetMe conference. However, channel
drivers that present audio with a varying rate will experience degraded
performance with a denoiser attached. This parameter allows a channel joining
the conference to choose not to have a denoiser attached without having to
unload <literal>func_speex</literal>.</para>
</option>
<option name="o">
<para>Set talker optimization - treats talkers who aren't speaking as
being muted, meaning (a) No encode is done on transmission and (b)
Received audio that is not registered as talking is omitted causing no
buildup in background noise.</para>
</option>
<option name="p" hasparams="optional">
<para>Allow user to exit the conference by pressing <literal>#</literal> (default)
or any of the defined keys. Dial plan execution will continue at the next
priority following MeetMe. The key used is set to channel variable
<variable>MEETME_EXIT_KEY</variable>.</para>
<argument name="keys" required="true" />
<note>
<para>Option <literal>s</literal> has priority for <literal>*</literal>
since it cannot change its activation code.</para>
</note>
</option>
<option name="P">
<para>Always prompt for the pin even if it is specified.</para>
</option>
<option name="q">
<para>Quiet mode (don't play enter/leave sounds).</para>
</option>
<option name="r">
<para>Record conference (records as <variable>MEETME_RECORDINGFILE</variable>
using format <variable>MEETME_RECORDINGFORMAT</variable>. Default filename is
<literal>meetme-conf-rec-${CONFNO}-${UNIQUEID}</literal> and the default format is
wav.</para>
</option>
<option name="s">
<para>Present menu (user or admin) when <literal>*</literal> is received
(send to menu).</para>
</option>
<option name="t">
<para>Set talk only mode. (Talk only, no listening).</para>
</option>
<option name="T">
<para>Set talker detection (sent to manager interface and meetme list).</para>
</option>
<option name="v" hasparams="optional">
<para>Announce when a user is joining or leaving the conference. Use the voicemail greeting as the announcement.
If the i or I options are set, the application will fall back to them if no voicemail greeting can be found.</para>
<argument name="mailbox@[context]" required="true">
<para>The mailbox and voicemail context to play from. If no context provided, assumed context is default.</para>
</argument>
</option>
<option name="w" hasparams="optional">
<para>Wait until the marked user enters the conference.</para>
<argument name="secs" required="true" />
</option>
<option name="x">
<para>Leave the conference when the last marked user leaves.</para>
</option>
<option name="X">
<para>Allow user to exit the conference by entering a valid single digit
extension <variable>MEETME_EXIT_CONTEXT</variable> or the current context
if that variable is not defined.</para>
<note>
<para>Option <literal>s</literal> has priority for <literal>*</literal>
since it cannot change its activation code.</para>
</note>
</option>
<option name="1">
<para>Do not play message when first person enters</para>
</option>
<option name="S">
<para>Kick the user <replaceable>x</replaceable> seconds <emphasis>after</emphasis> he entered into
the conference.</para>
<argument name="x" required="true" />
</option>
<option name="L" argsep=":">
<para>Limit the conference to <replaceable>x</replaceable> ms. Play a warning when
<replaceable>y</replaceable> ms are left. Repeat the warning every <replaceable>z</replaceable> ms.
The following special variables can be used with this option:</para>
<variablelist>
<variable name="CONF_LIMIT_TIMEOUT_FILE">
<para>File to play when time is up.</para>
</variable>
<variable name="CONF_LIMIT_WARNING_FILE">
<para>File to play as warning if <replaceable>y</replaceable> is defined. The
default is to say the time remaining.</para>
</variable>
</variablelist>
<argument name="x" />
<argument name="y" />
<argument name="z" />
</option>
</optionlist>
</parameter>
<parameter name="pin" />
</syntax>
<description>
<para>Enters the user into a specified MeetMe conference. If the <replaceable>confno</replaceable>
is omitted, the user will be prompted to enter one. User can exit the conference by hangup, or
if the <literal>p</literal> option is specified, by pressing <literal>#</literal>.</para>
<note>
<para>The DAHDI kernel modules and a functional DAHDI timing source (see dahdi_test)
must be present for conferencing to operate properly. In addition, the chan_dahdi channel driver
must be loaded for the <literal>i</literal> and <literal>r</literal> options to operate at
all.</para>
</note>
</description>
<see-also>
<ref type="application">MeetMeCount</ref>
<ref type="application">MeetMeAdmin</ref>
<ref type="application">MeetMeChannelAdmin</ref>
</see-also>
</application>
<application name="MeetMeCount" language="en_US">
<synopsis>
MeetMe participant count.
</synopsis>
<syntax>
<parameter name="confno" required="true">
<para>Conference number.</para>
</parameter>
<parameter name="var" />
</syntax>
<description>
<para>Plays back the number of users in the specified MeetMe conference.
If <replaceable>var</replaceable> is specified, playback will be skipped and the value
will be returned in the variable. Upon application completion, MeetMeCount will hangup
the channel, unless priority <literal>n+1</literal> exists, in which case priority progress will
continue.</para>
</description>
<see-also>
<ref type="application">MeetMe</ref>
</see-also>
</application>
<application name="MeetMeAdmin" language="en_US">
<synopsis>
MeetMe conference administration.
</synopsis>
<syntax>
<parameter name="confno" required="true" />
<parameter name="command" required="true">
<optionlist>
<option name="e">
<para>Eject last user that joined.</para>
</option>
<option name="E">
<para>Extend conference end time, if scheduled.</para>
</option>
<option name="k">
<para>Kick one user out of conference.</para>
</option>
<option name="K">
<para>Kick all users out of conference.</para>
</option>
<option name="l">
<para>Unlock conference.</para>
</option>
<option name="L">
<para>Lock conference.</para>
</option>
<option name="m">
<para>Unmute one user.</para>
</option>
<option name="M">
<para>Mute one user.</para>
</option>
<option name="n">
<para>Unmute all users in the conference.</para>
</option>
<option name="N">
<para>Mute all non-admin users in the conference.</para>
</option>
<option name="r">
<para>Reset one user's volume settings.</para>
</option>
<option name="R">
<para>Reset all users volume settings.</para>
</option>
<option name="s">
<para>Lower entire conference speaking volume.</para>
</option>
<option name="S">
<para>Raise entire conference speaking volume.</para>
</option>
<option name="t">
<para>Lower one user's talk volume.</para>
</option>
<option name="T">
<para>Raise one user's talk volume.</para>
</option>
<option name="u">
<para>Lower one user's listen volume.</para>
</option>
<option name="U">
<para>Raise one user's listen volume.</para>
</option>
<option name="v">
<para>Lower entire conference listening volume.</para>
</option>
<option name="V">
<para>Raise entire conference listening volume.</para>
</option>
</optionlist>
</parameter>
<parameter name="user" />
</syntax>
<description>
<para>Run admin <replaceable>command</replaceable> for conference <replaceable>confno</replaceable>.</para>
<para>Will additionally set the variable <variable>MEETMEADMINSTATUS</variable> with one of
the following values:</para>
<variablelist>
<variable name="MEETMEADMINSTATUS">
<value name="NOPARSE">
Invalid arguments.
</value>
<value name="NOTFOUND">
User specified was not found.
</value>
<value name="FAILED">
Another failure occurred.
</value>
<value name="OK">
The operation was completed successfully.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">MeetMe</ref>
</see-also>
</application>
<application name="MeetMeChannelAdmin" language="en_US">
<synopsis>
MeetMe conference Administration (channel specific).
</synopsis>
<syntax>
<parameter name="channel" required="true" />
<parameter name="command" required="true">
<optionlist>
<option name="k">
<para>Kick the specified user out of the conference he is in.</para>
</option>
<option name="m">
<para>Unmute the specified user.</para>
</option>
<option name="M">
<para>Mute the specified user.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Run admin <replaceable>command</replaceable> for a specific
<replaceable>channel</replaceable> in any conference.</para>
</description>
</application>
<application name="SLAStation" language="en_US">
<synopsis>
Shared Line Appearance Station.
</synopsis>
<syntax>
<parameter name="station" required="true">
<para>Station name</para>
</parameter>
</syntax>
<description>
<para>This application should be executed by an SLA station. The argument depends
on how the call was initiated. If the phone was just taken off hook, then the argument
<replaceable>station</replaceable> should be just the station name. If the call was
initiated by pressing a line key, then the station name should be preceded by an underscore
and the trunk name associated with that line button.</para>
<para>For example: <literal>station1_line1</literal></para>
<para>On exit, this application will set the variable <variable>SLASTATION_STATUS</variable> to
one of the following values:</para>
<variablelist>
<variable name="SLASTATION_STATUS">
<value name="FAILURE" />
<value name="CONGESTION" />
<value name="SUCCESS" />
</variable>
</variablelist>
</description>
</application>
<application name="SLATrunk" language="en_US">
<synopsis>
Shared Line Appearance Trunk.
</synopsis>
<syntax>
<parameter name="trunk" required="true">
<para>Trunk name</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="M" hasparams="optional">
<para>Play back the specified MOH <replaceable>class</replaceable>
instead of ringing</para>
<argument name="class" required="true" />
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application should be executed by an SLA trunk on an inbound call. The channel calling
this application should correspond to the SLA trunk with the name <replaceable>trunk</replaceable>
that is being passed as an argument.</para>
<para>On exit, this application will set the variable <variable>SLATRUNK_STATUS</variable> to
one of the following values:</para>
<variablelist>
<variable name="SLATRUNK_STATUS">
<value name="FAILURE" />
<value name="SUCCESS" />
<value name="UNANSWERED" />
<value name="RINGTIMEOUT" />
</variable>
</variablelist>
</description>
</application>
<function name="MEETME_INFO" language="en_US">
<synopsis>
Query a given conference of various properties.
</synopsis>
<syntax>
<parameter name="keyword" required="true">
<para>Options:</para>
<enumlist>
<enum name="lock">
<para>Boolean of whether the corresponding conference is locked.</para>
</enum>
<enum name="parties">
<para>Number of parties in a given conference</para>
</enum>
<enum name="activity">
<para>Duration of conference in seconds.</para>
</enum>
<enum name="dynamic">
<para>Boolean of whether the corresponding conference is dynamic.</para>
</enum>
</enumlist>
</parameter>
<parameter name="confno" required="true">
<para>Conference number to retrieve information from.</para>
</parameter>
</syntax>
<description />
<see-also>
<ref type="application">MeetMe</ref>
<ref type="application">MeetMeCount</ref>
<ref type="application">MeetMeAdmin</ref>
<ref type="application">MeetMeChannelAdmin</ref>
</see-also>
</function>
<manager name="MeetmeMute" language="en_US">
<synopsis>
Mute a Meetme user.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Meetme" required="true" />
<parameter name="Usernum" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="MeetmeUnmute" language="en_US">
<synopsis>
Unmute a Meetme user.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Meetme" required="true" />
<parameter name="Usernum" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="MeetmeList" language="en_US">
<synopsis>
List participants in a conference.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Conference" required="false">
<para>Conference number.</para>
</parameter>
</syntax>
<description>
<para>Lists all users in a particular MeetMe conference.
MeetmeList will follow as separate events, followed by a final event called
MeetmeListComplete.</para>
</description>
</manager>
<manager name="MeetmeListRooms" language="en_US">
<synopsis>
List active conferences.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Lists data about all active conferences.
MeetmeListRooms will follow as separate events, followed by a final event called
MeetmeListRoomsComplete.</para>
</description>
</manager>
<managerEvent language="en_US" name="MeetmeJoin">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a user joins a MeetMe conference.</synopsis>
<syntax>
<parameter name="Meetme">
<para>The identifier for the MeetMe conference.</para>
</parameter>
<parameter name="Usernum">
<para>The identifier of the MeetMe user who joined.</para>
</parameter>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MeetmeLeave</ref>
<ref type="application">MeetMe</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MeetmeLeave">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a user leaves a MeetMe conference.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeJoin']/managerEventInstance/syntax/parameter)" />
<channel_snapshot/>
<parameter name="Duration">
<para>The length of time in seconds that the Meetme user was in the conference.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">MeetmeJoin</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MeetmeEnd">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a MeetMe conference ends.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeJoin']/managerEventInstance/syntax/parameter[@name='Meetme'])" />
</syntax>
<see-also>
<ref type="managerEvent">MeetmeJoin</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MeetmeTalkRequest">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a MeetMe user has started talking.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeJoin']/managerEventInstance/syntax/parameter)" />
<channel_snapshot/>
<parameter name="Duration">
<para>The length of time in seconds that the Meetme user has been in the conference at the time of this event.</para>
</parameter>
<parameter name="Status">
<enumlist>
<enum name="on"/>
<enum name="off"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MeetmeTalking">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a MeetMe user begins or ends talking.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeJoin']/managerEventInstance/syntax/parameter)" />
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeTalkRequest']/managerEventInstance/syntax/parameter)" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MeetmeMute">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a MeetMe user is muted or unmuted.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeJoin']/managerEventInstance/syntax/parameter)" />
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='MeetmeTalkRequest']/managerEventInstance/syntax/parameter)" />
</syntax>
</managerEventInstance>
</managerEvent>
<application name="Pickup" language="en_US">
<synopsis>
Directed extension call pickup.
</synopsis>
<syntax>
<parameter name="targets" argsep="&amp;">
<argument name="extension" argsep="@" required="true">
<para>Specification of the pickup target.</para>
<argument name="extension" required="true"/>
<argument name="context" />
</argument>
<argument name="extension2" argsep="@" multiple="true">
<para>Additional specifications of pickup targets.</para>
<argument name="extension2" required="true"/>
<argument name="context2"/>
</argument>
</parameter>
</syntax>
<description>
<para>This application can pickup a specified ringing channel. The channel
to pickup can be specified in the following ways.</para>
<para>1) If no <replaceable>extension</replaceable> targets are specified,
the application will pickup a channel matching the pickup group of the
requesting channel.</para>
<para>2) If the <replaceable>extension</replaceable> is specified with a
<replaceable>context</replaceable> of the special string
<literal>PICKUPMARK</literal> (for example 10@PICKUPMARK), the application
will pickup a channel which has defined the channel variable
<variable>PICKUPMARK</variable> with the same value as
<replaceable>extension</replaceable> (in this example,
<literal>10</literal>).</para>
<para>3) If the <replaceable>extension</replaceable> is specified
with or without a <replaceable>context</replaceable>, the channel with a
matching <replaceable>extension</replaceable> and <replaceable>context</replaceable>
will be picked up. If no <replaceable>context</replaceable> is specified,
the current context will be used.</para>
<note>
<para>The <replaceable>extension</replaceable> is typically set on
matching channels by the dial application that created the channel. The
<replaceable>context</replaceable> is set on matching channels by the
channel driver for the device.</para>
</note>
</description>
</application>
<application name="PickupChan" language="en_US">
<synopsis>
Pickup a ringing channel.
</synopsis>
<syntax >
<parameter name="channel" argsep="&amp;" required="true">
<argument name="channel" required="true" />
<argument name="channel2" required="false" multiple="true" />
<para>List of channel names or channel uniqueids to pickup if ringing.
For example, a channel name could be <literal>SIP/bob</literal> or
<literal>SIP/bob-00000000</literal> to find
<literal>SIP/bob-00000000</literal>.
</para>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="p">
<para>Supplied channel names are prefixes. For example,
<literal>SIP/bob</literal> will match
<literal>SIP/bob-00000000</literal> and
<literal>SIP/bobby-00000000</literal>.
</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Pickup a specified <replaceable>channel</replaceable> if ringing.</para>
</description>
</application>
<application name="ICES" language="en_US">
<synopsis>
Encode and stream using 'ices'.
</synopsis>
<syntax>
<parameter name="config" required="true">
<para>ICES configuration file.</para>
</parameter>
</syntax>
<description>
<para>Streams to an icecast server using ices (available separately).
A configuration file must be supplied for ices (see contrib/asterisk-ices.xml).</para>
<note>
<para>ICES version 2 client and server required.</para>
</note>
</description>
</application>
<application name="System" language="en_US">
<synopsis>
Execute a system command.
</synopsis>
<syntax>
<parameter name="command" required="true">
<para>Command to execute</para>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</parameter>
</syntax>
<description>
<para>Executes a command by using system(). If the command
fails, the console should report a fallthrough.</para>
<para>Result of execution is returned in the <variable>SYSTEMSTATUS</variable> channel variable:</para>
<variablelist>
<variable name="SYSTEMSTATUS">
<value name="FAILURE">
Could not execute the specified command.
</value>
<value name="SUCCESS">
Specified command successfully executed.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="TrySystem" language="en_US">
<synopsis>
Try executing a system command.
</synopsis>
<syntax>
<parameter name="command" required="true">
<para>Command to execute</para>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</parameter>
</syntax>
<description>
<para>Executes a command by using system().</para>
<para>Result of execution is returned in the <variable>SYSTEMSTATUS</variable> channel variable:</para>
<variablelist>
<variable name="SYSTEMSTATUS">
<value name="FAILURE">
Could not execute the specified command.
</value>
<value name="SUCCESS">
Specified command successfully executed.
</value>
<value name="APPERROR">
Specified command successfully executed, but returned error code.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="Macro" language="en_US">
<synopsis>
Macro Implementation.
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the macro</para>
</parameter>
<parameter name="args">
<argument name="arg1" required="true" />
<argument name="arg2" multiple="true" />
</parameter>
</syntax>
<description>
<para>Executes a macro using the context macro-<replaceable>name</replaceable>,
jumping to the <literal>s</literal> extension of that context and executing each step,
then returning when the steps end.</para>
<para>The calling extension, context, and priority are stored in <variable>MACRO_EXTEN</variable>,
<variable>MACRO_CONTEXT</variable> and <variable>MACRO_PRIORITY</variable> respectively. Arguments
become <variable>ARG1</variable>, <variable>ARG2</variable>, etc in the macro context.</para>
<para>If you Goto out of the Macro context, the Macro will terminate and control will be returned
at the location of the Goto.</para>
<para>If <variable>MACRO_OFFSET</variable> is set at termination, Macro will attempt to continue
at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.</para>
<warning>
<para>Because of the way Macro is implemented (it executes the priorities contained within
it via sub-engine), and a fixed per-thread memory stack allowance, macros are limited to 7 levels
of nesting (macro calling macro calling macro, etc.); It may be possible that stack-intensive
applications in deeply nested macros could cause asterisk to crash earlier than this limit.
It is advised that if you need to deeply nest macro calls, that you use the Gosub application
(now allows arguments like a Macro) with explict Return() calls instead.</para>
</warning>
<warning>
<para>Use of the application <literal>WaitExten</literal> within a macro will not function
as expected. Please use the <literal>Read</literal> application in order to read DTMF from a channel
currently executing a macro.</para>
</warning>
</description>
<see-also>
<ref type="application">MacroExit</ref>
<ref type="application">Goto</ref>
<ref type="application">Gosub</ref>
</see-also>
</application>
<application name="MacroIf" language="en_US">
<synopsis>
Conditional Macro implementation.
</synopsis>
<syntax argsep="?">
<parameter name="expr" required="true" />
<parameter name="destination" required="true" argsep=":">
<argument name="macroiftrue" required="true">
<argument name="macroiftrue" required="true" />
<argument name="arg1" multiple="true" />
</argument>
<argument name="macroiffalse">
<argument name="macroiffalse" required="true" />
<argument name="arg1" multiple="true" />
</argument>
</parameter>
</syntax>
<description>
<para>Executes macro defined in <replaceable>macroiftrue</replaceable> if
<replaceable>expr</replaceable> is true (otherwise <replaceable>macroiffalse</replaceable>
if provided)</para>
<para>Arguments and return values as in application Macro()</para>
<xi:include xpointer="xpointer(/docs/application[@name='Macro']/description/warning[2])" />
</description>
<see-also>
<ref type="application">GotoIf</ref>
<ref type="application">GosubIf</ref>
<ref type="function">IF</ref>
</see-also>
</application>
<application name="MacroExclusive" language="en_US">
<synopsis>
Exclusive Macro Implementation.
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the macro</para>
</parameter>
<parameter name="arg1" />
<parameter name="arg2" multiple="true" />
</syntax>
<description>
<para>Executes macro defined in the context macro-<replaceable>name</replaceable>.
Only one call at a time may run the macro. (we'll wait if another call is busy
executing in the Macro)</para>
<para>Arguments and return values as in application Macro()</para>
<xi:include xpointer="xpointer(/docs/application[@name='Macro']/description/warning[2])" />
</description>
<see-also>
<ref type="application">Macro</ref>
</see-also>
</application>
<application name="MacroExit" language="en_US">
<synopsis>
Exit from Macro.
</synopsis>
<syntax />
<description>
<para>Causes the currently running macro to exit as if it had
ended normally by running out of priorities to execute.
If used outside a macro, will likely cause unexpected behavior.</para>
</description>
<see-also>
<ref type="application">Macro</ref>
</see-also>
</application>
<application name="WaitUntil" language="en_US">
<synopsis>
Wait (sleep) until the current time is the given epoch.
</synopsis>
<syntax>
<parameter name="epoch" required="true" />
</syntax>
<description>
<para>Waits until the given <replaceable>epoch</replaceable>.</para>
<para>Sets <variable>WAITUNTILSTATUS</variable> to one of the following values:</para>
<variablelist>
<variable name="WAITUNTILSTATUS">
<value name="OK">
Wait succeeded.
</value>
<value name="FAILURE">
Invalid argument.
</value>
<value name="HANGUP">
Channel hungup before time elapsed.
</value>
<value name="PAST">
Time specified had already past.
</value>
</variable>
</variablelist>
</description>
</application>
<application name="ExternalIVR" language="en_US">
<synopsis>
Interfaces with an external IVR application.
</synopsis>
<syntax>
<parameter name="command|ivr://host" required="true" hasparams="true">
<argument name="arg1" />
<argument name="arg2" multiple="yes" />
</parameter>
<parameter name="options">
<optionlist>
<option name="n">
<para>Tells ExternalIVR() not to answer the channel.</para>
</option>
<option name="i">
<para>Tells ExternalIVR() not to send a hangup and exit when the
channel receives a hangup, instead it sends an <literal>I</literal>
informative message meaning that the external application MUST hang
up the call with an <literal>H</literal> command.</para>
</option>
<option name="d">
<para>Tells ExternalIVR() to run on a channel that has been hung up
and will not look for hangups. The external application must exit with
an <literal>E</literal> command.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Either forks a process to run given command or makes a socket to connect
to given host and starts a generator on the channel. The generator's play list
is controlled by the external application, which can add and clear entries via
simple commands issued over its stdout. The external application will receive
all DTMF events received on the channel, and notification if the channel is
hung up. The received on the channel, and notification if the channel is hung
up. The application will not be forcibly terminated when the channel is hung up.
For more information see <filename>doc/AST.pdf</filename>.</para>
</description>
</application>
<application name="AMD" language="en_US">
<synopsis>
Attempt to detect answering machines.
</synopsis>
<syntax>
<parameter name="initialSilence" required="false">
<para>Is maximum initial silence duration before greeting.</para>
<para>If this is exceeded, the result is detection as a MACHINE</para>
</parameter>
<parameter name="greeting" required="false">
<para>is the maximum length of a greeting.</para>
<para>If this is exceeded, the result is detection as a MACHINE</para>
</parameter>
<parameter name="afterGreetingSilence" required="false">
<para>Is the silence after detecting a greeting.</para>
<para>If this is exceeded, the result is detection as a HUMAN</para>
</parameter>
<parameter name="totalAnalysis Time" required="false">
<para>Is the maximum time allowed for the algorithm</para>
<para>to decide on whether the audio represents a HUMAN, or a MACHINE</para>
</parameter>
<parameter name="miniumWordLength" required="false">
<para>Is the minimum duration of Voice considered to be a word</para>
</parameter>
<parameter name="betweenWordSilence" required="false">
<para>Is the minimum duration of silence after a word to
consider the audio that follows to be a new word</para>
</parameter>
<parameter name="maximumNumberOfWords" required="false">
<para>Is the maximum number of words in a greeting</para>
<para>If this is REACHED, then the result is detection as a MACHINE</para>
</parameter>
<parameter name="silenceThreshold" required="false">
<para>What is the average level of noise from 0 to 32767 which if not exceeded, should be considered silence?</para>
</parameter>
<parameter name="maximumWordLength" required="false">
<para>Is the maximum duration of a word to accept.</para>
<para>If exceeded, then the result is detection as a MACHINE</para>
</parameter>
</syntax>
<description>
<para>This application attempts to detect answering machines at the beginning
of outbound calls. Simply call this application after the call
has been answered (outbound only, of course).</para>
<para>When loaded, AMD reads amd.conf and uses the parameters specified as
default values. Those default values get overwritten when the calling AMD
with parameters.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="AMDSTATUS">
<para>This is the status of the answering machine detection</para>
<value name="MACHINE" />
<value name="HUMAN" />
<value name="NOTSURE" />
<value name="HANGUP" />
</variable>
<variable name="AMDCAUSE">
<para>Indicates the cause that led to the conclusion</para>
<value name="TOOLONG">
Total Time.
</value>
<value name="INITIALSILENCE">
Silence Duration - Initial Silence.
</value>
<value name="HUMAN">
Silence Duration - afterGreetingSilence.
</value>
<value name="LONGGREETING">
Voice Duration - Greeting.
</value>
<value name="MAXWORDLENGTH">
Word Length - max length of a single word.
</value>
<value name="MAXWORDS">
Word Count - maximum number of words.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">WaitForSilence</ref>
<ref type="application">WaitForNoise</ref>
</see-also>
</application>
<application name="AlarmReceiver" language="en_US">
<synopsis>
Provide support for receiving alarm reports from a burglar or fire alarm panel.
</synopsis>
<syntax />
<description>
<para>This application should be called whenever there is an alarm panel calling in to dump its events.
The application will handshake with the alarm panel, and receive events, validate them, handshake them,
and store them until the panel hangs up. Once the panel hangs up, the application will run the system
command specified by the eventcmd setting in <filename>alarmreceiver.conf</filename> and pipe the
events to the standard input of the application.
The configuration file also contains settings for DTMF timing, and for the loudness of the
acknowledgement tones.</para>
<note>
<para>Few Ademco DTMF signalling formats are detected automaticaly: Contact ID, Express 4+1,
Express 4+2, High Speed and Super Fast.</para>
</note>
<para>The application is affected by the following variables:</para>
<variablelist>
<variable name="ALARMRECEIVER_CALL_LIMIT">
<para>Maximum call time, in milliseconds.</para>
<para>If set, this variable causes application to exit after the specified time.</para>
</variable>
<variable name="ALARMRECEIVER_RETRIES_LIMIT">
<para>Maximum number of retries per call.</para>
<para>If set, this variable causes application to exit after the specified number of messages.</para>
</variable>
</variablelist>
</description>
<see-also>
<ref type="filename">alarmreceiver.conf</ref>
</see-also>
</application>
<application name="DAHDIRAS" language="en_US">
<synopsis>
Executes DAHDI ISDN RAS application.
</synopsis>
<syntax>
<parameter name="args" required="true">
<para>A list of parameters to pass to the pppd daemon,
separated by <literal>,</literal> characters.</para>
</parameter>
</syntax>
<description>
<para>Executes a RAS server using pppd on the given channel.
The channel must be a clear channel (i.e. PRI source) and a DAHDI
channel to be able to use this function (No modem emulation is included).</para>
<para>Your pppd must be patched to be DAHDI aware.</para>
</description>
</application>
<application name="DBdel" language="en_US">
<synopsis>
Delete a key from the asterisk database.
</synopsis>
<syntax argsep="/">
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>This application will delete a <replaceable>key</replaceable> from the Asterisk
database.</para>
<note>
<para>This application has been DEPRECATED in favor of the DB_DELETE function.</para>
</note>
</description>
<see-also>
<ref type="function">DB_DELETE</ref>
<ref type="application">DBdeltree</ref>
<ref type="function">DB</ref>
</see-also>
</application>
<application name="DBdeltree" language="en_US">
<synopsis>
Delete a family or keytree from the asterisk database.
</synopsis>
<syntax argsep="/">
<parameter name="family" required="true" />
<parameter name="keytree" />
</syntax>
<description>
<para>This application will delete a <replaceable>family</replaceable> or <replaceable>keytree</replaceable>
from the Asterisk database.</para>
</description>
<see-also>
<ref type="function">DB_DELETE</ref>
<ref type="application">DBdel</ref>
<ref type="function">DB</ref>
</see-also>
</application>
<application name="Gosub" language="en_US">
<synopsis>
Jump to label, saving return address.
</synopsis>
<syntax>
<parameter name="context" />
<parameter name="exten" />
<parameter name="priority" required="true" hasparams="optional">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</parameter>
</syntax>
<description>
<para>Jumps to the label specified, saving the return address.</para>
</description>
<see-also>
<ref type="application">GosubIf</ref>
<ref type="application">Macro</ref>
<ref type="application">Goto</ref>
<ref type="application">Return</ref>
<ref type="application">StackPop</ref>
</see-also>
</application>
<application name="GosubIf" language="en_US">
<synopsis>
Conditionally jump to label, saving return address.
</synopsis>
<syntax argsep="?">
<parameter name="condition" required="true" />
<parameter name="destination" required="true" argsep=":">
<argument name="labeliftrue" hasparams="optional">
<para>Continue at <replaceable>labeliftrue</replaceable> if the condition is true.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
<argument name="arg1" required="true" multiple="true" />
<argument name="argN" />
</argument>
<argument name="labeliffalse" hasparams="optional">
<para>Continue at <replaceable>labeliffalse</replaceable> if the condition is false.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
<argument name="arg1" required="true" multiple="true" />
<argument name="argN" />
</argument>
</parameter>
</syntax>
<description>
<para>If the condition is true, then jump to labeliftrue. If false, jumps to
labeliffalse, if specified. In either case, a jump saves the return point
in the dialplan, to be returned to with a Return.</para>
</description>
<see-also>
<ref type="application">Gosub</ref>
<ref type="application">Return</ref>
<ref type="application">MacroIf</ref>
<ref type="function">IF</ref>
<ref type="application">GotoIf</ref>
<ref type="application">Goto</ref>
</see-also>
</application>
<application name="Return" language="en_US">
<synopsis>
Return from gosub routine.
</synopsis>
<syntax>
<parameter name="value">
<para>Return value.</para>
</parameter>
</syntax>
<description>
<para>Jumps to the last label on the stack, removing it. The return <replaceable>value</replaceable>, if
any, is saved in the channel variable <variable>GOSUB_RETVAL</variable>.</para>
</description>
<see-also>
<ref type="application">Gosub</ref>
<ref type="application">StackPop</ref>
</see-also>
</application>
<application name="StackPop" language="en_US">
<synopsis>
Remove one address from gosub stack.
</synopsis>
<syntax />
<description>
<para>Removes last label on the stack, discarding it.</para>
</description>
<see-also>
<ref type="application">Return</ref>
<ref type="application">Gosub</ref>
</see-also>
</application>
<function name="LOCAL" language="en_US">
<synopsis>
Manage variables local to the gosub stack frame.
</synopsis>
<syntax>
<parameter name="varname" required="true" />
</syntax>
<description>
<para>Read and write a variable local to the gosub stack frame, once we Return() it will be lost
(or it will go back to whatever value it had before the Gosub()).</para>
</description>
<see-also>
<ref type="application">Gosub</ref>
<ref type="application">GosubIf</ref>
<ref type="application">Return</ref>
</see-also>
</function>
<function name="LOCAL_PEEK" language="en_US">
<synopsis>
Retrieve variables hidden by the local gosub stack frame.
</synopsis>
<syntax>
<parameter name="n" required="true" />
<parameter name="varname" required="true" />
</syntax>
<description>
<para>Read a variable <replaceable>varname</replaceable> hidden by
<replaceable>n</replaceable> levels of gosub stack frames. Note that ${LOCAL_PEEK(0,foo)}
is the same as <variable>foo</variable>, since the value of <replaceable>n</replaceable>
peeks under 0 levels of stack frames; in other words, 0 is the current level. If
<replaceable>n</replaceable> exceeds the available number of stack frames, then an empty
string is returned.</para>
</description>
<see-also>
<ref type="application">Gosub</ref>
<ref type="application">GosubIf</ref>
<ref type="application">Return</ref>
</see-also>
</function>
<function name="STACK_PEEK" language="en_US">
<synopsis>
View info about the location which called Gosub
</synopsis>
<syntax>
<parameter name="n" required="true" />
<parameter name="which" required="true" />
<parameter name="suppress" required="false" />
</syntax>
<description>
<para>Read the calling <literal>c</literal>ontext, <literal>e</literal>xtension,
<literal>p</literal>riority, or <literal>l</literal>abel, as specified by
<replaceable>which</replaceable>, by going up <replaceable>n</replaceable> frames
in the Gosub stack. If <replaceable>suppress</replaceable> is true, then if the
number of available stack frames is exceeded, then no error message will be
printed.</para>
</description>
</function>
<agi name="gosub" language="en_US">
<synopsis>
Cause the channel to execute the specified dialplan subroutine.
</synopsis>
<syntax>
<parameter name="context" required="true" />
<parameter name="extension" required="true" />
<parameter name="priority" required="true" />
<parameter name="optional-argument" />
</syntax>
<description>
<para>Cause the channel to execute the specified dialplan subroutine,
returning to the dialplan with execution of a Return().</para>
</description>
<see-also>
<ref type="application">GoSub</ref>
</see-also>
</agi>
<managerEvent language="en_US" name="VarSet">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>Raised when a variable local to the gosub stack frame is set due to a subroutine call.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Variable">
<para>The LOCAL variable being set.</para>
<note>
<para>The variable name will always be enclosed with
<literal>LOCAL()</literal>
</para>
</note>
</parameter>
<parameter name="Value">
<para>The new value of the variable.</para>
</parameter>
</syntax>
<see-also>
<ref type="application">GoSub</ref>
<ref type="agi">gosub</ref>
<ref type="function">LOCAL</ref>
<ref type="function">LOCAL_PEEK</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<application name="WaitForRing" language="en_US">
<synopsis>
Wait for Ring Application.
</synopsis>
<syntax>
<parameter name="timeout" required="true" />
</syntax>
<description>
<para>Returns <literal>0</literal> after waiting at least <replaceable>timeout</replaceable> seconds,
and only after the next ring has completed. Returns <literal>0</literal> on success or
<literal>-1</literal> on hangup.</para>
</description>
</application>
<application name="SMS" language="en_US">
<synopsis>
Communicates with SMS service centres and SMS capable analogue phones.
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the queue used in <filename>/var/spool/asterisk/sms</filename></para>
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Answer, i.e. send initial FSK packet.</para>
</option>
<option name="s">
<para>Act as service centre talking to a phone.</para>
</option>
<option name="t">
<para>Use protocol 2 (default used is protocol 1).</para>
</option>
<option name="p">
<para>Set the initial delay to N ms (default is <literal>300</literal>).
addr and body are a deprecated format to send messages out.</para>
</option>
<option name="r">
<para>Set the Status Report Request (SRR) bit.</para>
</option>
<option name="o">
<para>The body should be coded as octets not 7-bit symbols.</para>
</option>
</optionlist>
</parameter>
<parameter name="addr" />
<parameter name="body" />
</syntax>
<description>
<para>SMS handles exchange of SMS data with a call to/from SMS capable phone or SMS PSTN service center.
Can send and/or receive SMS messages. Works to ETSI ES 201 912; compatible with BT SMS PSTN service in
UK and Telecom Italia in Italy.</para>
<para>Typical usage is to use to handle calls from the SMS service centre CLI, or to set up a call using
<literal>outgoing</literal> or manager interface to connect service centre to SMS().</para>
<para>"Messages are processed as per text file message queues. smsq (a separate software) is a command to
generate message queues and send messages.</para>
<note>
<para>The protocol has tight delay bounds. Please use short frames and disable/keep short the
jitter buffer on the ATA to make sure that respones (ACK etc.) are received in time.</para>
</note>
</description>
</application>
<managerEvent language="en_US" name="Cdr">
<managerEventInstance class="EVENT_FLAG_CDR">
<synopsis>Raised when a CDR is generated.</synopsis>
<syntax>
<parameter name="AccountCode">
<para>The account code of the Party A channel.</para>
</parameter>
<parameter name="Source">
<para>The Caller ID number associated with the Party A in the CDR.</para>
</parameter>
<parameter name="Destination">
<para>The dialplan extension the Party A was executing.</para>
</parameter>
<parameter name="DestinationContext">
<para>The dialplan context the Party A was executing.</para>
</parameter>
<parameter name="CallerID">
<para>The Caller ID name associated with the Party A in the CDR.</para>
</parameter>
<parameter name="Channel">
<para>The channel name of the Party A.</para>
</parameter>
<parameter name="DestinationChannel">
<para>The channel name of the Party B.</para>
</parameter>
<parameter name="LastApplication">
<para>The last dialplan application the Party A executed.</para>
</parameter>
<parameter name="LastData">
<para>
The parameters passed to the last dialplan application the
Party A executed.
</para>
</parameter>
<parameter name="StartTime">
<para>The time the CDR was created.</para>
</parameter>
<parameter name="AnswerTime">
<para>
The earliest of either the time when Party A answered, or
the start time of this CDR.
</para>
</parameter>
<parameter name="EndTime">
<para>
The time when the CDR was finished. This occurs when the
Party A hangs up or when the bridge between Party A and
Party B is broken.
</para>
</parameter>
<parameter name="Duration">
<para>The time, in seconds, of <replaceable>EndTime</replaceable> - <replaceable>StartTime</replaceable>.</para>
</parameter>
<parameter name="BillableSeconds">
<para>The time, in seconds, of <replaceable>AnswerTime</replaceable> - <replaceable>StartTime</replaceable>.</para>
</parameter>
<parameter name="Disposition">
<para>The final known disposition of the CDR.</para>
<enumlist>
<enum name="NO ANSWER">
<para>The channel was not answered. This is the default disposition.</para>
</enum>
<enum name="FAILED">
<para>The channel attempted to dial but the call failed.</para>
<note>
<para>The congestion setting in <filename>cdr.conf</filename> can result
in the <literal>AST_CAUSE_CONGESTION</literal> hang up cause or the
<literal>CONGESTION</literal> dial status to map to this disposition.
</para>
</note>
</enum>
<enum name="BUSY">
<para>The channel attempted to dial but the remote party was busy.</para>
</enum>
<enum name="ANSWERED">
<para>The channel was answered. The hang up cause will no longer
impact the disposition of the CDR.</para>
</enum>
<enum name="CONGESTION">
<para>The channel attempted to dial but the remote party was congested.</para>
</enum>
</enumlist>
</parameter>
<parameter name="AMAFlags">
<para>A flag that informs a billing system how to treat the CDR.</para>
<enumlist>
<enum name="OMIT">
<para>This CDR should be ignored.</para>
</enum>
<enum name="BILLING">
<para>This CDR contains valid billing data.</para>
</enum>
<enum name="DOCUMENTATION">
<para>This CDR is for documentation purposes.</para>
</enum>
</enumlist>
</parameter>
<parameter name="UniqueID">
<para>A unique identifier for the Party A channel.</para>
</parameter>
<parameter name="UserField">
<para>
A user defined field set on the channels. If set on both the Party A
and Party B channel, the userfields of both are concatenated and
separated by a <literal>;</literal>.
</para>
</parameter>
</syntax>
<description>
<para>
The <replaceable>Cdr</replaceable> event is only raised when the
<filename>cdr_manager</filename> backend is loaded and registered with
the CDR engine.
</para>
<note>
<para>
This event can contain additional fields depending on the configuration
provided by <filename>cdr_manager.conf</filename>.
</para>
</note>
</description>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="CEL">
<managerEventInstance class="EVENT_FLAG_CEL">
<synopsis>Raised when a Channel Event Log is generated for a channel.</synopsis>
<syntax>
<parameter name="EventName">
<para>
The name of the CEL event being raised. This can include
both the system defined CEL events, as well as user defined
events.
</para>
<note>
<para>All events listed here may not be raised, depending
on the configuration in <filename>cel.conf</filename>.</para>
</note>
<enumlist>
<enum name="CHAN_START">
<para>A channel was created.</para>
</enum>
<enum name="CHAN_END">
<para>A channel was terminated.</para>
</enum>
<enum name="ANSWER">
<para>A channel answered.</para>
</enum>
<enum name="HANGUP">
<para>A channel was hung up.</para>
</enum>
<enum name="BRIDGE_ENTER">
<para>A channel entered a bridge.</para>
</enum>
<enum name="BRIDGE_EXIT">
<para>A channel left a bridge.</para>
</enum>
<enum name="APP_START">
<para>A channel entered into a tracked application.</para>
</enum>
<enum name="APP_END">
<para>A channel left a tracked application.</para>
</enum>
<enum name="PARK_START">
<para>A channel was parked.</para>
</enum>
<enum name="PARK_END">
<para>A channel was unparked.</para>
</enum>
<enum name="BLINDTRANSFER">
<para>A channel initiated a blind transfer.</para>
</enum>
<enum name="ATTENDEDTRANSFER">
<para>A channel initiated an attended transfer.</para>
</enum>
<enum name="PICKUP">
<para>A channel initated a call pickup.</para>
</enum>
<enum name="FORWARD">
<para>A channel is being forwarded to another destination.</para>
</enum>
<enum name="LINKEDID_END">
<para>The linked ID associated with this channel is being retired.</para>
</enum>
<enum name="LOCAL_OPTIMIZE">
<para>A Local channel optimization has occurred.</para>
</enum>
<enum name="USER_DEFINED">
<para>A user defined type.</para>
<note>
<para>
This event is only present if <literal>show_user_defined</literal>
in <filename>cel.conf</filename> is <literal>True</literal>. Otherwise,
the user defined event will be placed directly in the
<replaceable>EventName</replaceable> field.
</para>
</note>
</enum>
</enumlist>
</parameter>
<parameter name="AccountCode">
<para>The channel's account code.</para>
</parameter>
<parameter name="CallerIDnum">
<para>The Caller ID number.</para>
</parameter>
<parameter name="CallerIDname">
<para>The Caller ID name.</para>
</parameter>
<parameter name="CallerIDani">
<para>The Caller ID Automatic Number Identification.</para>
</parameter>
<parameter name="CallerIDrdnis">
<para>The Caller ID Redirected Dialed Number Identification Service.</para>
</parameter>
<parameter name="CallerIDdnid">
<para>The Caller ID Dialed Number Identifier.</para>
</parameter>
<parameter name="Exten">
<para>The dialplan extension the channel is currently executing in.</para>
</parameter>
<parameter name="Context">
<para>The dialplan context the channel is currently executing in.</para>
</parameter>
<parameter name="Application">
<para>The dialplan application the channel is currently executing.</para>
</parameter>
<parameter name="AppData">
<para>The arguments passed to the dialplan <replaceable>Application</replaceable>.</para>
</parameter>
<parameter name="EventTime">
<para>The time the CEL event occurred.</para>
</parameter>
<parameter name="AMAFlags">
<para>A flag that informs a billing system how to treat the CEL.</para>
<enumlist>
<enum name="OMIT">
<para>This event should be ignored.</para>
</enum>
<enum name="BILLING">
<para>This event contains valid billing data.</para>
</enum>
<enum name="DOCUMENTATION">
<para>This event is for documentation purposes.</para>
</enum>
</enumlist>
</parameter>
<parameter name="UniqueID">
<para>The unique ID of the channel.</para>
</parameter>
<parameter name="LinkedID">
<para>The linked ID of the channel, which ties this event to other related channel's events.</para>
</parameter>
<parameter name="UserField">
<para>
A user defined field set on a channel, containing arbitrary
application specific data.
</para>
</parameter>
<parameter name="Peer">
<para>
If this channel is in a bridge, the channel that it is in
a bridge with.
</para>
</parameter>
<parameter name="PeerAccount">
<para>
If this channel is in a bridge, the accountcode of the
channel it is in a bridge with.
</para>
</parameter>
<parameter name="Extra">
<para>
Some events will have event specific data that accompanies the CEL record.
This extra data is JSON encoded, and is dependent on the event in
question.
</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<function name="CURL" language="en_US">
<synopsis>
Retrieve content from a remote web or ftp server
</synopsis>
<syntax>
<parameter name="url" required="true" />
<parameter name="post-data">
<para>If specified, an <literal>HTTP POST</literal> will be
performed with the content of
<replaceable>post-data</replaceable>, instead of an
<literal>HTTP GET</literal> (default).</para>
</parameter>
</syntax>
<description />
<see-also>
<ref type="function">CURLOPT</ref>
</see-also>
</function>
<function name="CURLOPT" language="en_US">
<synopsis>
Sets various options for future invocations of CURL.
</synopsis>
<syntax>
<parameter name="key" required="yes">
<enumlist>
<enum name="cookie">
<para>A cookie to send with the request. Multiple
cookies are supported.</para>
</enum>
<enum name="conntimeout">
<para>Number of seconds to wait for a connection to succeed</para>
</enum>
<enum name="dnstimeout">
<para>Number of seconds to wait for DNS to be resolved</para>
</enum>
<enum name="ftptext">
<para>For FTP URIs, force a text transfer (boolean)</para>
</enum>
<enum name="ftptimeout">
<para>For FTP URIs, number of seconds to wait for a
server response</para>
</enum>
<enum name="header">
<para>Include header information in the result
(boolean)</para>
</enum>
<enum name="httptimeout">
<para>For HTTP(S) URIs, number of seconds to wait for a
server response</para>
</enum>
<enum name="maxredirs">
<para>Maximum number of redirects to follow</para>
</enum>
<enum name="proxy">
<para>Hostname or IP address to use as a proxy server</para>
</enum>
<enum name="proxytype">
<para>Type of <literal>proxy</literal></para>
<enumlist>
<enum name="http" />
<enum name="socks4" />
<enum name="socks5" />
</enumlist>
</enum>
<enum name="proxyport">
<para>Port number of the <literal>proxy</literal></para>
</enum>
<enum name="proxyuserpwd">
<para>A <replaceable>username</replaceable>
<literal>:</literal>
<replaceable>password</replaceable>
combination to use for authenticating requests through a
<literal>proxy</literal>
</para>
</enum>
<enum name="referer">
<para>Referer URL to use for the request</para>
</enum>
<enum name="useragent">
<para>UserAgent string to use for the request</para>
</enum>
<enum name="userpwd">
<para>A <replaceable>username</replaceable>
<literal>:</literal>
<replaceable>password</replaceable>
to use for authentication when the server response to
an initial request indicates a 401 status code.</para>
</enum>
<enum name="ssl_verifypeer">
<para>Whether to verify the server certificate against
a list of known root certificate authorities (boolean).</para>
</enum>
<enum name="hashcompat">
<para>Assuming the responses will be in <literal>key1=value1&amp;key2=value2</literal>
format, reformat the response such that it can be used
by the <literal>HASH</literal> function.</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
<enum name="legacy">
<para>Also translate <literal>+</literal> to the
space character, in violation of current RFC
standards.</para>
</enum>
</enumlist>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Options may be set globally or per channel. Per-channel
settings will override global settings.</para>
</description>
<see-also>
<ref type="function">CURL</ref>
<ref type="function">HASH</ref>
</see-also>
</function>
<function name="HANGUPCAUSE" language="en_US">
<synopsis>
Gets per-channel hangupcause information from the channel.
</synopsis>
<syntax>
<parameter name="channel" required="true">
<para>The name of the channel for which to retrieve cause information.</para>
</parameter>
<parameter name="type" required="true">
<para>Parameter describing which type of information is requested. Types are:</para>
<enumlist>
<enum name="tech">
<para>Technology-specific cause information</para>
</enum>
<enum name="ast">
<para>Translated Asterisk cause code</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Gets technology-specific or translated Asterisk cause code information
from the channel for the specified channel that resulted from a dial.</para>
</description>
<see-also>
<ref type="function">HANGUPCAUSE_KEYS</ref>
<ref type="application">HangupCauseClear</ref>
</see-also>
</function>
<function name="HANGUPCAUSE_KEYS" language="en_US">
<synopsis>
Gets the list of channels for which hangup causes are available.
</synopsis>
<description>
<para>Returns a comma-separated list of channel names to be used with the HANGUPCAUSE function.</para>
</description>
<see-also>
<ref type="function">HANGUPCAUSE</ref>
<ref type="application">HangupCauseClear</ref>
</see-also>
</function>
<application name="HangupCauseClear" language="en_US">
<synopsis>
Clears hangup cause information from the channel that is available through HANGUPCAUSE.
</synopsis>
<description>
<para>Clears all channel-specific hangup cause information from the channel.
This is never done automatically (i.e. for new Dial()s).</para>
</description>
<see-also>
<ref type="function">HANGUPCAUSE</ref>
<ref type="function">HANGUPCAUSE_KEYS</ref>
</see-also>
</application>
<function name="GROUP_COUNT" language="en_US">
<synopsis>
Counts the number of channels in the specified group.
</synopsis>
<syntax argsep="@">
<parameter name="groupname">
<para>Group name.</para>
</parameter>
<parameter name="category">
<para>Category name</para>
</parameter>
</syntax>
<description>
<para>Calculates the group count for the specified group, or uses the
channel's current group if not specified (and non-empty).</para>
</description>
</function>
<function name="GROUP_MATCH_COUNT" language="en_US">
<synopsis>
Counts the number of channels in the groups matching the specified pattern.
</synopsis>
<syntax argsep="@">
<parameter name="groupmatch" required="true">
<para>A standard regular expression used to match a group name.</para>
</parameter>
<parameter name="category">
<para>A standard regular expression used to match a category name.</para>
</parameter>
</syntax>
<description>
<para>Calculates the group count for all groups that match the specified pattern.
Note: category matching is applied after matching based on group.
Uses standard regular expression matching on both (see regex(7)).</para>
</description>
</function>
<function name="GROUP" language="en_US">
<synopsis>
Gets or sets the channel group.
</synopsis>
<syntax>
<parameter name="category">
<para>Category name.</para>
</parameter>
</syntax>
<description>
<para>
<replaceable>category</replaceable> can be employed for more fine grained group management. Each channel
can only be member of exactly one group per <replaceable>category</replaceable>.</para>
</description>
</function>
<function name="GROUP_LIST" language="en_US">
<synopsis>
Gets a list of the groups set on a channel.
</synopsis>
<syntax />
<description>
<para>Gets a list of the groups set on a channel.</para>
</description>
</function>
<function name="EXTENSION_STATE" language="en_US">
<synopsis>
Get an extension's state.
</synopsis>
<syntax argsep="@">
<parameter name="extension" required="true" />
<parameter name="context">
<para>If it is not specified defaults to <literal>default</literal>.</para>
</parameter>
</syntax>
<description>
<para>The EXTENSION_STATE function can be used to retrieve the state from any
hinted extension. For example:</para>
<para>NoOp(1234@default has state ${EXTENSION_STATE(1234)})</para>
<para>NoOp(4567@home has state ${EXTENSION_STATE(4567@home)})</para>
<para>The possible values returned by this function are:</para>
<para>UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING |
RINGINUSE | HOLDINUSE | ONHOLD</para>
</description>
</function>
<function name="MD5" language="en_US">
<synopsis>
Computes an MD5 digest.
</synopsis>
<syntax>
<parameter name="data" required="true" />
</syntax>
<description>
<para>Computes an MD5 digest.</para>
</description>
</function>
<function name="JITTERBUFFER" language="en_US">
<synopsis>
Add a Jitterbuffer to the Read side of the channel. This dejitters the audio stream before it reaches the Asterisk core. This is a write only function.
</synopsis>
<syntax>
<parameter name="jitterbuffer type" required="true">
<optionlist>
<option name="fixed">
<para>Set a fixed jitterbuffer on the channel.</para>
</option>
<option name="adaptive">
<para>Set an adaptive jitterbuffer on the channel.</para>
</option>
<option name="disabled">
<para>Remove a previously set jitterbuffer from the channel.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Jitterbuffers are constructed in two different ways.
The first always take three arguments: <replaceable>max_size</replaceable>,
<replaceable>resync_threshold</replaceable>, and <replaceable>target_extra</replaceable>.
Alternatively, a single argument of <literal>default</literal> can be provided,
which will construct the default jitterbuffer for the given
<replaceable>jitterbuffer type</replaceable>.</para>
<para>The arguments are:</para>
<para>
<replaceable>max_size</replaceable>: Length in milliseconds of the buffer.
Defaults to 200 ms.</para>
<para>
<replaceable>resync_threshold</replaceable>: The length in milliseconds over
which a timestamp difference will result in resyncing the jitterbuffer.
Defaults to 1000ms.</para>
<para>target_extra: This option only affects the adaptive jitterbuffer. It represents
the amount time in milliseconds by which the new jitter buffer will pad its size.
Defaults to 40ms.</para>
<example title="Fixed with defaults" language="text">
exten => 1,1,Set(JITTERBUFFER(fixed)=default)
</example>
<example title="Fixed with 200ms max size" language="text">
exten => 1,1,Set(JITTERBUFFER(fixed)=200)
</example>
<example title="Fixed with 200ms max size, resync threshold 1500" language="text">
exten => 1,1,Set(JITTERBUFFER(fixed)=200,1500)
</example>
<example title="Adaptive with defaults" language="text">
exten => 1,1,Set(JITTERBUFFER(adaptive)=default)
</example>
<example title="Adaptive with 200ms max size, 60ms target extra" language="text">
exten => 1,1,Set(JITTERBUFFER(adaptive)=200,,60)
</example>
<example title="Set a fixed jitterbuffer with defaults; then remove it" language="text">
exten => 1,1,Set(JITTERBUFFER(fixed)=default)
exten => 1,n,Set(JITTERBUFFER(disabled)=)
</example>
<note>
<para>If a channel specifies a jitterbuffer due to channel driver configuration and
the JITTERBUFFER function has set a jitterbuffer for that channel, the jitterbuffer set by
the JITTERBUFFER function will take priority and the jitterbuffer set by the channel
configuration will not be applied.</para>
</note>
</description>
</function>
<function name="TIMEOUT" language="en_US">
<synopsis>
Gets or sets timeouts on the channel. Timeout values are in seconds.
</synopsis>
<syntax>
<parameter name="timeouttype" required="true">
<para>The timeout that will be manipulated. The possible timeout types
are: <literal>absolute</literal>, <literal>digit</literal> or
<literal>response</literal>
</para>
</parameter>
</syntax>
<description>
<para>The timeouts that can be manipulated are:</para>
<para>
<literal>absolute</literal>: The absolute maximum amount of time permitted for a call.
Setting of 0 disables the timeout.</para>
<para>
<literal>digit</literal>: The maximum amount of time permitted between digits when the
user is typing in an extension. When this timeout expires,
after the user has started to type in an extension, the
extension will be considered complete, and will be
interpreted. Note that if an extension typed in is valid,
it will not have to timeout to be tested, so typically at
the expiry of this timeout, the extension will be considered
invalid (and thus control would be passed to the <literal>i</literal>
extension, or if it doesn't exist the call would be
terminated). The default timeout is 5 seconds.</para>
<para>
<literal>response</literal>: The maximum amount of time permitted after falling through a
series of priorities for a channel in which the user may
begin typing an extension. If the user does not type an
extension in this amount of time, control will pass to the
<literal>t</literal> extension if it exists, and if not the call would be
terminated. The default timeout is 10 seconds.</para>
</description>
</function>
<function name="AST_CONFIG" language="en_US">
<synopsis>
Retrieve a variable from a configuration file.
</synopsis>
<syntax>
<parameter name="config_file" required="true" />
<parameter name="category" required="true" />
<parameter name="variable_name" required="true" />
<parameter name="index" required="false">
<para>If there are multiple variables with the same name, you can specify
<literal>0</literal> for the first item (default), <literal>-1</literal> for the last
item, or any other number for that specific item. <literal>-1</literal> is useful
when the variable is derived from a template and you want the effective value (the last
occurrence), not the value from the template (the first occurrence).</para>
</parameter>
</syntax>
<description>
<para>This function reads a variable from an Asterisk configuration file.</para>
</description>
</function>
<function name="TALK_DETECT" language="en_US">
<synopsis>
Raises notifications when Asterisk detects silence or talking on a channel.
</synopsis>
<syntax>
<parameter name="action" required="true">
<optionlist>
<option name="remove">
<para>W/O. Remove talk detection from the channel.</para>
</option>
<option name="set">
<para>W/O. Enable TALK_DETECT and/or configure talk detection
parameters. Can be called multiple times to change parameters
on a channel with talk detection already enabled.</para>
<argument name="dsp_silence_threshold" required="false">
<para>The time in milliseconds before which a user is considered silent.</para>
</argument>
<argument name="dsp_talking_threshold" required="false">
<para>The time in milliseconds after which a user is considered talking.</para>
</argument>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The TALK_DETECT function enables events on the channel
it is applied to. These events can be emited over AMI, ARI, and
potentially other Asterisk modules that listen for the internal
notification.</para>
<para>The function has two parameters that can optionally be passed
when <literal>set</literal> on a channel: <replaceable>dsp_talking_threshold</replaceable>
and <replaceable>dsp_silence_threshold</replaceable>.</para>
<para>
<replaceable>dsp_talking_threshold</replaceable> is the time in milliseconds of sound
above what the dsp has established as base line silence for a user
before a user is considered to be talking. By default, the value of
<replaceable>silencethreshold</replaceable> from <filename>dsp.conf</filename>
is used. If this value is set too tight events may be
falsely triggered by variants in room noise.</para>
<para>Valid values are 1 through 2^31.</para>
<para>
<replaceable>dsp_silence_threshold</replaceable> is the time in milliseconds of sound
falling within what the dsp has established as baseline silence before
a user is considered be silent. If this value is set too low events
indicating the user has stopped talking may get falsely sent out when
the user briefly pauses during mid sentence.</para>
<para>The best way to approach this option is to set it slightly above
the maximum amount of ms of silence a user may generate during
natural speech.</para>
<para>By default this value is 2500ms. Valid values are 1
through 2^31.</para>
<para>Example:</para>
<para>same => n,Set(TALK_DETECT(set)=) ; Enable talk detection</para>
<para>same => n,Set(TALK_DETECT(set)=1200) ; Update existing talk detection's silence threshold to 1200 ms</para>
<para>same => n,Set(TALK_DETECT(remove)=) ; Remove talk detection</para>
<para>same => n,Set(TALK_DETECT(set)=,128) ; Enable and set talk threshold to 128</para>
<para>This function will set the following variables:</para>
<note>
<para>The TALK_DETECT function uses an audiohook to inspect the
voice media frames on a channel. Other functions, such as JITTERBUFFER,
DENOISE, and AGC use a similar mechanism. Audiohooks are processed
in the order in which they are placed on the channel. As such,
it typically makes sense to place functions that modify the voice
media data prior to placing the TALK_DETECT function, as this will
yield better results.</para>
<para>Example:</para>
<para>same => n,Set(DENOISE(rx)=on) ; Denoise received audio</para>
<para>same => n,Set(TALK_DETECT(set)=) ; Perform talk detection on the denoised received audio</para>
</note>
</description>
</function>
<function name="ISNULL" language="en_US">
<synopsis>
Check if a value is NULL.
</synopsis>
<syntax>
<parameter name="data" required="true" />
</syntax>
<description>
<para>Returns <literal>1</literal> if NULL or <literal>0</literal> otherwise.</para>
</description>
</function>
<function name="SET" language="en_US">
<synopsis>
SET assigns a value to a channel variable.
</synopsis>
<syntax argsep="=">
<parameter name="varname" required="true" />
<parameter name="value" />
</syntax>
<description>
</description>
</function>
<function name="EXISTS" language="en_US">
<synopsis>
Test the existence of a value.
</synopsis>
<syntax>
<parameter name="data" required="true" />
</syntax>
<description>
<para>Returns <literal>1</literal> if exists, <literal>0</literal> otherwise.</para>
</description>
</function>
<function name="IF" language="en_US">
<synopsis>
Check for an expresion.
</synopsis>
<syntax argsep="?">
<parameter name="expresion" required="true" />
<parameter name="retvalue" argsep=":" required="true">
<argument name="true" />
<argument name="false" />
</parameter>
</syntax>
<description>
<para>Returns the data following <literal>?</literal> if true, else the data following <literal>:</literal></para>
</description>
</function>
<function name="IFTIME" language="en_US">
<synopsis>
Temporal Conditional.
</synopsis>
<syntax argsep="?">
<parameter name="timespec" required="true" />
<parameter name="retvalue" required="true" argsep=":">
<argument name="true" />
<argument name="false" />
</parameter>
</syntax>
<description>
<para>Returns the data following <literal>?</literal> if true, else the data following <literal>:</literal></para>
</description>
</function>
<function name="IMPORT" language="en_US">
<synopsis>
Retrieve the value of a variable from another channel.
</synopsis>
<syntax>
<parameter name="channel" required="true" />
<parameter name="variable" required="true" />
</syntax>
<description>
</description>
</function>
<function name = "AUDIOHOOK_INHERIT" language="en_US">
<synopsis>
DEPRECATED: Used to set whether an audiohook may be inherited to another
channel. Due to architectural changes in Asterisk 12, audiohook inheritance
is performed automatically and this function now lacks function.
</synopsis>
<description>
<para>Prior to Asterisk 12, masquerades would occur under all sorts of
situations which were hard to predict. In Asterisk 12, masquerades only
occur as a result of a small set of operations for which inheriting all
audiohooks from the original channel is now safe. So in Asterisk 12.5+,
all audiohooks are inherited without needing other controls expressing
which audiohooks should be inherited under which conditions.</para>
</description>
</function>
<function name="VOLUME" language="en_US">
<synopsis>
Set the TX or RX volume of a channel.
</synopsis>
<syntax>
<parameter name="direction" required="true">
<para>Must be <literal>TX</literal> or <literal>RX</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="p">
<para>Enable DTMF volume control</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The VOLUME function can be used to increase or decrease the <literal>tx</literal> or
<literal>rx</literal> gain of any channel.</para>
<para>For example:</para>
<para>Set(VOLUME(TX)=3)</para>
<para>Set(VOLUME(RX)=2)</para>
<para>Set(VOLUME(TX,p)=3)</para>
<para>Set(VOLUME(RX,p)=3)</para>
</description>
</function>
<function name="PRESENCE_STATE" language="en_US">
<synopsis>
Get or Set a presence state.
</synopsis>
<syntax>
<parameter name="provider" required="true">
<para>The provider of the presence, such as <literal>CustomPresence</literal></para>
</parameter>
<parameter name="field" required="true">
<para>Which field of the presence state information is wanted.</para>
<optionlist>
<option name="value">
<para>The current presence, such as <literal>away</literal></para>
</option>
<option name="subtype">
<para>Further information about the current presence</para>
</option>
<option name="message">
<para>A custom message that may indicate further details about the presence</para>
</option>
</optionlist>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="e">
<para>On Write - Use this option when the subtype and message provided are Base64
encoded. The values will be stored encoded within Asterisk, but all consumers of
the presence state (e.g. the SIP presence event package) will receive decoded values.</para>
<para>On Read - Retrieves unencoded message/subtype in Base64 encoded form.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The PRESENCE_STATE function can be used to retrieve the presence from any
presence provider. For example:</para>
<para>NoOp(SIP/mypeer has presence ${PRESENCE_STATE(SIP/mypeer,value)})</para>
<para>NoOp(Conference number 1234 has presence message ${PRESENCE_STATE(MeetMe:1234,message)})</para>
<para>The PRESENCE_STATE function can also be used to set custom presence state from
the dialplan. The <literal>CustomPresence:</literal> prefix must be used. For example:</para>
<para>Set(PRESENCE_STATE(CustomPresence:lamp1)=away,temporary,Out to lunch)</para>
<para>Set(PRESENCE_STATE(CustomPresence:lamp2)=dnd,,Trying to get work done)</para>
<para>Set(PRESENCE_STATE(CustomPresence:lamp3)=xa,T24gdmFjYXRpb24=,,e)</para>
<para>Set(BASE64_LAMP3_PRESENCE=${PRESENCE_STATE(CustomPresence:lamp3,subtype,e)})</para>
<para>You can subscribe to the status of a custom presence state using a hint in
the dialplan:</para>
<para>exten => 1234,hint,,CustomPresence:lamp1</para>
<para>The possible values for both uses of this function are:</para>
<para>not_set | unavailable | available | away | xa | chat | dnd</para>
</description>
</function>
<function name="REALTIME" language="en_US">
<synopsis>
RealTime Read/Write Functions.
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="fieldmatch" required="true" />
<parameter name="matchvalue" />
<parameter name="delim1|field">
<para>Use <replaceable>delim1</replaceable> with <replaceable>delim2</replaceable> on
read and <replaceable>field</replaceable> without <replaceable>delim2</replaceable> on
write</para>
<para>If we are reading and <replaceable>delim1</replaceable> is not specified, defaults
to <literal>,</literal>
</para>
</parameter>
<parameter name="delim2">
<para>Parameter only used when reading, if not specified defaults to <literal>=</literal></para>
</parameter>
</syntax>
<description>
<para>This function will read or write values from/to a RealTime repository.
REALTIME(....) will read names/values from the repository, and
REALTIME(....)= will write a new value/field to the repository. On a
read, this function returns a delimited text string. The name/value
pairs are delimited by <replaceable>delim1</replaceable>, and the name and value are delimited
between each other with delim2.
If there is no match, NULL will be returned by the function.
On a write, this function will always return NULL.</para>
</description>
<see-also>
<ref type="function">REALTIME_STORE</ref>
<ref type="function">REALTIME_DESTROY</ref>
<ref type="function">REALTIME_FIELD</ref>
<ref type="function">REALTIME_HASH</ref>
</see-also>
</function>
<function name="REALTIME_STORE" language="en_US">
<synopsis>
RealTime Store Function.
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="field1" required="true" />
<parameter name="fieldN" required="true" multiple="true" />
<parameter name="field30" required="true" />
</syntax>
<description>
<para>This function will insert a new set of values into the RealTime repository.
If RT engine provides an unique ID of the stored record, REALTIME_STORE(...)=..
creates channel variable named RTSTOREID, which contains value of unique ID.
Currently, a maximum of 30 field/value pairs is supported.</para>
</description>
<see-also>
<ref type="function">REALTIME</ref>
<ref type="function">REALTIME_DESTROY</ref>
<ref type="function">REALTIME_FIELD</ref>
<ref type="function">REALTIME_HASH</ref>
</see-also>
</function>
<function name="REALTIME_DESTROY" language="en_US">
<synopsis>
RealTime Destroy Function.
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="fieldmatch" required="true" />
<parameter name="matchvalue" />
<parameter name="delim1" />
<parameter name="delim2" />
</syntax>
<description>
<para>This function acts in the same way as REALTIME(....) does, except that
it destroys the matched record in the RT engine.</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be read from the
dialplan, and not directly from external protocols. It can, however, be
executed as a write operation (<literal>REALTIME_DESTROY(family, fieldmatch)=ignored</literal>)</para>
</note>
</description>
<see-also>
<ref type="function">REALTIME</ref>
<ref type="function">REALTIME_STORE</ref>
<ref type="function">REALTIME_FIELD</ref>
<ref type="function">REALTIME_HASH</ref>
</see-also>
</function>
<function name="REALTIME_FIELD" language="en_US">
<synopsis>
RealTime query function.
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="fieldmatch" required="true" />
<parameter name="matchvalue" required="true" />
<parameter name="fieldname" required="true" />
</syntax>
<description>
<para>This function retrieves a single item, <replaceable>fieldname</replaceable>
from the RT engine, where <replaceable>fieldmatch</replaceable> contains the value
<replaceable>matchvalue</replaceable>. When written to, the REALTIME_FIELD() function
performs identically to the REALTIME() function.</para>
</description>
<see-also>
<ref type="function">REALTIME</ref>
<ref type="function">REALTIME_STORE</ref>
<ref type="function">REALTIME_DESTROY</ref>
<ref type="function">REALTIME_HASH</ref>
</see-also>
</function>
<function name="REALTIME_HASH" language="en_US">
<synopsis>
RealTime query function.
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="fieldmatch" required="true" />
<parameter name="matchvalue" required="true" />
</syntax>
<description>
<para>This function retrieves a single record from the RT engine, where
<replaceable>fieldmatch</replaceable> contains the value
<replaceable>matchvalue</replaceable> and formats the output suitably, such that
it can be assigned to the HASH() function. The HASH() function then provides
a suitable method for retrieving each field value of the record.</para>
</description>
<see-also>
<ref type="function">REALTIME</ref>
<ref type="function">REALTIME_STORE</ref>
<ref type="function">REALTIME_DESTROY</ref>
<ref type="function">REALTIME_FIELD</ref>
</see-also>
</function>
<function name="ENUMQUERY" language="en_US">
<synopsis>
Initiate an ENUM query.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="method-type">
<para>If no <replaceable>method-type</replaceable> is given, the default will be
<literal>sip</literal>.</para>
</parameter>
<parameter name="zone-suffix">
<para>If no <replaceable>zone-suffix</replaceable> is given, the default will be
<literal>e164.arpa</literal>
</para>
</parameter>
</syntax>
<description>
<para>This will do a ENUM lookup of the given phone number.</para>
</description>
</function>
<function name="ENUMRESULT" language="en_US">
<synopsis>
Retrieve results from a ENUMQUERY.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The identifier returned by the ENUMQUERY function.</para>
</parameter>
<parameter name="resultnum" required="true">
<para>The number of the result that you want to retrieve.</para>
<para>Results start at <literal>1</literal>. If this argument is specified
as <literal>getnum</literal>, then it will return the total number of results
that are available or -1 on error.</para>
</parameter>
</syntax>
<description>
<para>This function will retrieve results from a previous use
of the ENUMQUERY function.</para>
</description>
</function>
<function name="ENUMLOOKUP" language="en_US">
<synopsis>
General or specific querying of NAPTR records for ENUM or ENUM-like DNS pointers.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="method-type">
<para>If no <replaceable>method-type</replaceable> is given, the default will be
<literal>sip</literal>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="c">
<para>Returns an integer count of the number of NAPTRs of a certain RR type.</para>
<para>Combination of <literal>c</literal> and Method-type of <literal>ALL</literal> will
return a count of all NAPTRs for the record or -1 on error.</para>
</option>
<option name="u">
<para>Returns the full URI and does not strip off the URI-scheme.</para>
</option>
<option name="s">
<para>Triggers ISN specific rewriting.</para>
</option>
<option name="i">
<para>Looks for branches into an Infrastructure ENUM tree.</para>
</option>
<option name="d">
<para>for a direct DNS lookup without any flipping of digits.</para>
</option>
</optionlist>
</parameter>
<parameter name="record#">
<para>If no <replaceable>record#</replaceable> is given,
defaults to <literal>1</literal>.</para>
</parameter>
<parameter name="zone-suffix">
<para>If no <replaceable>zone-suffix</replaceable> is given, the default will be
<literal>e164.arpa</literal>
</para>
</parameter>
</syntax>
<description>
<para>For more information see <filename>doc/AST.pdf</filename>.</para>
</description>
</function>
<function name="TXTCIDNAME" language="en_US">
<synopsis>
TXTCIDNAME looks up a caller name via DNS.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="zone-suffix">
<para>If no <replaceable>zone-suffix</replaceable> is given, the default will be
<literal>e164.arpa</literal>
</para>
</parameter>
</syntax>
<description>
<para>This function looks up the given phone number in DNS to retrieve
the caller id name. The result will either be blank or be the value
found in the TXT record in DNS.</para>
</description>
</function>
<function name="VERSION" language="en_US">
<synopsis>
Return the Version info for this Asterisk.
</synopsis>
<syntax>
<parameter name="info">
<para>The possible values are:</para>
<enumlist>
<enum name="ASTERISK_VERSION_NUM">
<para>A string of digits is returned, e.g. 10602 for 1.6.2 or 100300 for 10.3.0,
or 999999 when using an SVN build.</para>
</enum>
<enum name="BUILD_USER">
<para>The string representing the user's name whose account
was used to configure Asterisk, is returned.</para>
</enum>
<enum name="BUILD_HOSTNAME">
<para>The string representing the name of the host on which Asterisk was configured, is returned.</para>
</enum>
<enum name="BUILD_MACHINE">
<para>The string representing the type of machine on which Asterisk was configured, is returned.</para>
</enum>
<enum name="BUILD_OS">
<para>The string representing the OS of the machine on which Asterisk was configured, is returned.</para>
</enum>
<enum name="BUILD_DATE">
<para>The string representing the date on which Asterisk was configured, is returned.</para>
</enum>
<enum name="BUILD_KERNEL">
<para>The string representing the kernel version of the machine on which Asterisk
was configured, is returned.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>If there are no arguments, return the version of Asterisk in this format: SVN-branch-1.4-r44830M</para>
<para>Example: Set(junky=${VERSION()};</para>
<para>Sets junky to the string <literal>SVN-branch-1.6-r74830M</literal>, or possibly, <literal>SVN-trunk-r45126M</literal>.</para>
</description>
</function>
<function name="SRVQUERY" language="en_US">
<synopsis>
Initiate an SRV query.
</synopsis>
<syntax>
<parameter name="service" required="true">
<para>The service for which to look up SRV records. An example would be something
like <literal>_sip._udp.example.com</literal>
</para>
</parameter>
</syntax>
<description>
<para>This will do an SRV lookup of the given service.</para>
</description>
</function>
<function name="SRVRESULT" language="en_US">
<synopsis>
Retrieve results from an SRVQUERY.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The identifier returned by the SRVQUERY function.</para>
</parameter>
<parameter name="resultnum" required="true">
<para>The number of the result that you want to retrieve.</para>
<para>Results start at <literal>1</literal>. If this argument is specified
as <literal>getnum</literal>, then it will return the total number of results
that are available.</para>
</parameter>
</syntax>
<description>
<para>This function will retrieve results from a previous use
of the SRVQUERY function.</para>
</description>
</function>
<function name="DIALGROUP" language="en_US">
<synopsis>
Manages a group of users for dialing.
</synopsis>
<syntax>
<parameter name="group" required="true" />
<parameter name="op">
<para>The operation name, possible values are:</para>
<para>
<literal>add</literal> - add a channel name or interface (write-only)</para>
<para>
<literal>del</literal> - remove a channel name or interface (write-only)</para>
</parameter>
</syntax>
<description>
<para>Presents an interface meant to be used in concert with the Dial
application, by presenting a list of channels which should be dialled when
referenced.</para>
<para>When DIALGROUP is read from, the argument is interpreted as the particular
<replaceable>group</replaceable> for which a dial should be attempted. When DIALGROUP is written to
with no arguments, the entire list is replaced with the argument specified.</para>
<para>Functionality is similar to a queue, except that when no interfaces are
available, execution may continue in the dialplan. This is useful when
you want certain people to be the first to answer any calls, with immediate
fallback to a queue when the front line people are busy or unavailable, but
you still want front line people to log in and out of that group, just like
a queue.</para>
<para>Example:</para>
<para>exten => 1,1,Set(DIALGROUP(mygroup,add)=SIP/10)</para>
<para>exten => 1,n,Set(DIALGROUP(mygroup,add)=SIP/20)</para>
<para>exten => 1,n,Dial(${DIALGROUP(mygroup)})</para>
</description>
</function>
<function name="AGC" language="en_US">
<synopsis>
Apply automatic gain control to audio on a channel.
</synopsis>
<syntax>
<parameter name="channeldirection" required="true">
<para>This can be either <literal>rx</literal> or <literal>tx</literal></para>
</parameter>
</syntax>
<description>
<para>The AGC function will apply automatic gain control to the audio on the
channel that it is executed on. Using <literal>rx</literal> for audio received
and <literal>tx</literal> for audio transmitted to the channel. When using this
function you set a target audio level. It is primarily intended for use with
analog lines, but could be useful for other channels as well. The target volume
is set with a number between <literal>1-32768</literal>. The larger the number
the louder (more gain) the channel will receive.</para>
<para>Examples:</para>
<para>exten => 1,1,Set(AGC(rx)=8000)</para>
<para>exten => 1,2,Set(AGC(tx)=off)</para>
</description>
</function>
<function name="DENOISE" language="en_US">
<synopsis>
Apply noise reduction to audio on a channel.
</synopsis>
<syntax>
<parameter name="channeldirection" required="true">
<para>This can be either <literal>rx</literal> or <literal>tx</literal>
the values that can be set to this are either <literal>on</literal> and
<literal>off</literal>
</para>
</parameter>
</syntax>
<description>
<para>The DENOISE function will apply noise reduction to audio on the channel
that it is executed on. It is very useful for noisy analog lines, especially
when adjusting gains or using AGC. Use <literal>rx</literal> for audio received from the channel
and <literal>tx</literal> to apply the filter to the audio being sent to the channel.</para>
<para>Examples:</para>
<para>exten => 1,1,Set(DENOISE(rx)=on)</para>
<para>exten => 1,2,Set(DENOISE(tx)=off)</para>
</description>
</function>
<function name="ICONV" language="en_US">
<synopsis>
Converts charsets of strings.
</synopsis>
<syntax>
<parameter name="in-charset" required="true">
<para>Input charset</para>
</parameter>
<parameter name="out-charset" required="true">
<para>Output charset</para>
</parameter>
<parameter name="string" required="true">
<para>String to convert, from <replaceable>in-charset</replaceable> to <replaceable>out-charset</replaceable></para>
</parameter>
</syntax>
<description>
<para>Converts string from <replaceable>in-charset</replaceable> into <replaceable>out-charset</replaceable>.
For available charsets, use <literal>iconv -l</literal> on your shell command line.</para>
<note>
<para>Due to limitations within the API, ICONV will not currently work with
charsets with embedded NULLs. If found, the string will terminate.</para>
</note>
</description>
</function>
<function name="CHANNELS" language="en_US">
<synopsis>
Gets the list of channels, optionally filtering by a regular expression.
</synopsis>
<syntax>
<parameter name="regular_expression" />
</syntax>
<description>
<para>Gets the list of channels, optionally filtering by a <replaceable>regular_expression</replaceable>. If
no argument is provided, all known channels are returned. The
<replaceable>regular_expression</replaceable> must correspond to
the POSIX.2 specification, as shown in <emphasis>regex(7)</emphasis>. The list returned
will be space-delimited.</para>
</description>
</function>
<function name="MASTER_CHANNEL" language="en_US">
<synopsis>
Gets or sets variables on the master channel
</synopsis>
<description>
<para>Allows access to the channel which created the current channel, if any. If the channel is already
a master channel, then accesses local channel variables.</para>
</description>
</function>
<function name="CHANNEL" language="en_US">
<synopsis>
Gets/sets various pieces of information about the channel.
</synopsis>
<syntax>
<parameter name="item" required="true">
<para>Standard items (provided by all channel technologies) are:</para>
<enumlist>
<enum name="amaflags">
<para>R/W the Automatic Message Accounting (AMA) flags on the channel.
When read from a channel, the integer value will always be returned.
When written to a channel, both the string format or integer value
is accepted.</para>
<enumlist>
<enum name="1">
<para>
<literal>OMIT</literal>
</para>
</enum>
<enum name="2">
<para>
<literal>BILLING</literal>
</para>
</enum>
<enum name="3">
<para>
<literal>DOCUMENTATION</literal>
</para>
</enum>
</enumlist>
</enum>
<enum name="accountcode">
<para>R/W the channel's account code.</para>
</enum>
<enum name="audioreadformat">
<para>R/O format currently being read.</para>
</enum>
<enum name="audionativeformat">
<para>R/O format used natively for audio.</para>
</enum>
<enum name="audiowriteformat">
<para>R/O format currently being written.</para>
</enum>
<enum name="dtmf_features">
<para>R/W The channel's DTMF bridge features.
May include one or more of 'T' 'K' 'H' 'W' and 'X' in a similar manner to options
in the <literal>Dial</literal> application. When setting it, the features string
must be all upper case.</para>
</enum>
<enum name="callgroup">
<para>R/W numeric call pickup groups that this channel is a member.</para>
</enum>
<enum name="pickupgroup">
<para>R/W numeric call pickup groups this channel can pickup.</para>
</enum>
<enum name="namedcallgroup">
<para>R/W named call pickup groups that this channel is a member.</para>
</enum>
<enum name="namedpickupgroup">
<para>R/W named call pickup groups this channel can pickup.</para>
</enum>
<enum name="channeltype">
<para>R/O technology used for channel.</para>
</enum>
<enum name="checkhangup">
<para>R/O Whether the channel is hanging up (1/0)</para>
</enum>
<enum name="after_bridge_goto">
<para>R/W the parseable goto string indicating where the channel is
expected to return to in the PBX after exiting the next bridge it joins
on the condition that it doesn't hang up. The parseable goto string uses
the same syntax as the <literal>Goto</literal> application.</para>
</enum>
<enum name="hangup_handler_pop">
<para>W/O Replace the most recently added hangup handler
with a new hangup handler on the channel if supplied. The
assigned string is passed to the Gosub application when
the channel is hung up. Any optionally omitted context
and exten are supplied by the channel pushing the handler
before it is pushed.</para>
</enum>
<enum name="hangup_handler_push">
<para>W/O Push a hangup handler onto the channel hangup
handler stack. The assigned string is passed to the
Gosub application when the channel is hung up. Any
optionally omitted context and exten are supplied by the
channel pushing the handler before it is pushed.</para>
</enum>
<enum name="hangup_handler_wipe">
<para>W/O Wipe the entire hangup handler stack and replace
with a new hangup handler on the channel if supplied. The
assigned string is passed to the Gosub application when
the channel is hung up. Any optionally omitted context
and exten are supplied by the channel pushing the handler
before it is pushed.</para>
</enum>
<enum name="language">
<para>R/W language for sounds played.</para>
</enum>
<enum name="musicclass">
<para>R/W class (from musiconhold.conf) for hold music.</para>
</enum>
<enum name="name">
<para>The name of the channel</para>
</enum>
<enum name="parkinglot">
<para>R/W parkinglot for parking.</para>
</enum>
<enum name="rxgain">
<para>R/W set rxgain level on channel drivers that support it.</para>
</enum>
<enum name="secure_bridge_signaling">
<para>Whether or not channels bridged to this channel require secure signaling (1/0)</para>
</enum>
<enum name="secure_bridge_media">
<para>Whether or not channels bridged to this channel require secure media (1/0)</para>
</enum>
<enum name="state">
<para>R/O state of the channel</para>
</enum>
<enum name="tonezone">
<para>R/W zone for indications played</para>
</enum>
<enum name="transfercapability">
<para>R/W ISDN Transfer Capability, one of:</para>
<enumlist>
<enum name="SPEECH" />
<enum name="DIGITAL" />
<enum name="RESTRICTED_DIGITAL" />
<enum name="3K1AUDIO" />
<enum name="DIGITAL_W_TONES" />
<enum name="VIDEO" />
</enumlist>
</enum>
<enum name="txgain">
<para>R/W set txgain level on channel drivers that support it.</para>
</enum>
<enum name="videonativeformat">
<para>R/O format used natively for video</para>
</enum>
<enum name="trace">
<para>R/W whether or not context tracing is enabled, only available
<emphasis>if CHANNEL_TRACE is defined</emphasis>.</para>
</enum>
<enum name="hangupsource">
<para>R/W returns the channel responsible for hangup.</para>
</enum>
<enum name="appname">
<para>R/O returns the internal application name.</para>
</enum>
<enum name="appdata">
<para>R/O returns the application data if available.</para>
</enum>
<enum name="exten">
<para>R/O returns the extension for an outbound channel.</para>
</enum>
<enum name="context">
<para>R/O returns the context for an outbound channel.</para>
</enum>
<enum name="channame">
<para>R/O returns the channel name for an outbound channel.</para>
</enum>
<enum name="uniqueid">
<para>R/O returns the channel uniqueid.</para>
</enum>
<enum name="linkedid">
<para>R/O returns the linkedid if available, otherwise returns the uniqueid.</para>
</enum>
<enum name="max_forwards">
<para>R/W The maximum number of forwards allowed.</para>
</enum>
<enum name="callid">
<para>R/O Call identifier log tag associated with the channel
e.g., <literal>[C-00000000]</literal>.</para>
</enum>
</enumlist>
<xi:include xpointer="xpointer(/docs/info[@name='CHANNEL'])" />
</parameter>
</syntax>
<description>
<para>Gets/sets various pieces of information about the channel, additional <replaceable>item</replaceable> may
be available from the channel driver; see its documentation for details. Any <replaceable>item</replaceable>
requested that is not available on the current channel will return an empty string.</para>
<example title="Standard CHANNEL item examples">
; Push a hangup handler subroutine existing at dialplan
; location default,s,1 onto the current channel
same => n,Set(CHANNEL(hangup_handler_push)=default,s,1)
; Set the current tonezone to Germany (de)
same => n,Set(CHANNEL(tonezone)=de)
; Set the allowed maximum number of forwarding attempts
same => n,Set(CHANNEL(max_forwards)=10)
; If this channel is ejected from its next bridge, and if
; the channel is not hung up, begin executing dialplan at
; location default,after-bridge,1
same => n,Set(CHANNEL(after_bridge_goto)=default,after-bridge,1)
; Log the current state of the channel
same => n,Log(NOTICE, This channel is: ${CHANNEL(state)})
</example>
<xi:include xpointer="xpointer(/docs/info[@name='CHANNEL_EXAMPLES'])" />
</description>
</function>
<function name="FRAME_TRACE" language="en_US">
<synopsis>
View internal ast_frames as they are read and written on a channel.
</synopsis>
<syntax>
<parameter name="filter list type" required="true">
<para>A filter can be applied to the trace to limit what frames are viewed. This
filter can either be a <literal>white</literal> or <literal>black</literal> list
of frame types. When no filter type is present, <literal>white</literal> is
used. If no arguments are provided at all, all frames will be output.
</para>
<para>Below are the different types of frames that can be filtered.</para>
<enumlist>
<enum name = "DTMF_BEGIN" />
<enum name = "DTMF_END" />
<enum name = "VOICE" />
<enum name = "VIDEO" />
<enum name = "CONTROL" />
<enum name = "NULL" />
<enum name = "IAX" />
<enum name = "TEXT" />
<enum name = "IMAGE" />
<enum name = "HTML" />
<enum name = "CNG" />
<enum name = "MODEM" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Examples:</para>
<para>exten => 1,1,Set(FRAME_TRACE(white)=DTMF_BEGIN,DTMF_END); view only DTMF frames. </para>
<para>exten => 1,1,Set(FRAME_TRACE()=DTMF_BEGIN,DTMF_END); view only DTMF frames. </para>
<para>exten => 1,1,Set(FRAME_TRACE(black)=DTMF_BEGIN,DTMF_END); view everything except DTMF frames. </para>
</description>
</function>
<function name="ODBC_FETCH" language="en_US">
<synopsis>
Fetch a row from a multirow query.
</synopsis>
<syntax>
<parameter name="result-id" required="true" />
</syntax>
<description>
<para>For queries which are marked as mode=multirow, the original
query returns a <replaceable>result-id</replaceable> from which results
may be fetched. This function implements the actual fetch of the results.</para>
<para>This also sets <variable>ODBC_FETCH_STATUS</variable>.</para>
<variablelist>
<variable name="ODBC_FETCH_STATUS">
<value name="SUCESS">
If rows are available.
</value>
<value name="FAILURE">
If no rows are available.
</value>
</variable>
</variablelist>
</description>
</function>
<application name="ODBCFinish" language="en_US">
<synopsis>
Clear the resultset of a sucessful multirow query.
</synopsis>
<syntax>
<parameter name="result-id" required="true" />
</syntax>
<description>
<para>For queries which are marked as mode=multirow, this will clear
any remaining rows of the specified resultset.</para>
</description>
</application>
<function name="SQL_ESC" language="en_US">
<synopsis>
Escapes single ticks for use in SQL statements.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Used in SQL templates to escape data which may contain single ticks
<literal>'</literal> which are otherwise used to delimit data.</para>
<para>Example: SELECT foo FROM bar WHERE baz='${SQL_ESC(${ARG1})}'</para>
</description>
</function>
<function name="SPRINTF" language="en_US">
<synopsis>
Format a variable according to a format string.
</synopsis>
<syntax>
<parameter name="format" required="true" />
<parameter name="arg1" required="true" />
<parameter name="arg2" multiple="true" />
<parameter name="argN" />
</syntax>
<description>
<para>Parses the format string specified and returns a string matching
that format. Supports most options found in <emphasis>sprintf(3)</emphasis>.
Returns a shortened string if a format specifier is not recognized.</para>
</description>
<see-also>
<ref type="manpage">sprintf(3)</ref>
</see-also>
</function>
<function name="DB" language="en_US">
<synopsis>
Read from or write to the Asterisk database.
</synopsis>
<syntax argsep="/">
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>This function will read from or write a value to the Asterisk database. On a
read, this function returns the corresponding value from the database, or blank
if it does not exist. Reading a database value will also set the variable
DB_RESULT. If you wish to find out if an entry exists, use the DB_EXISTS
function.</para>
</description>
<see-also>
<ref type="application">DBdel</ref>
<ref type="function">DB_DELETE</ref>
<ref type="application">DBdeltree</ref>
<ref type="function">DB_EXISTS</ref>
</see-also>
</function>
<function name="DB_EXISTS" language="en_US">
<synopsis>
Check to see if a key exists in the Asterisk database.
</synopsis>
<syntax argsep="/">
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>This function will check to see if a key exists in the Asterisk
database. If it exists, the function will return <literal>1</literal>. If not,
it will return <literal>0</literal>. Checking for existence of a database key will
also set the variable DB_RESULT to the key's value if it exists.</para>
</description>
<see-also>
<ref type="function">DB</ref>
</see-also>
</function>
<function name="DB_KEYS" language="en_US">
<synopsis>
Obtain a list of keys within the Asterisk database.
</synopsis>
<syntax>
<parameter name="prefix" />
</syntax>
<description>
<para>This function will return a comma-separated list of keys existing
at the prefix specified within the Asterisk database. If no argument is
provided, then a list of key families will be returned.</para>
</description>
</function>
<function name="DB_DELETE" language="en_US">
<synopsis>
Return a value from the database and delete it.
</synopsis>
<syntax argsep="/">
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>This function will retrieve a value from the Asterisk database
and then remove that key from the database. <variable>DB_RESULT</variable>
will be set to the key's value if it exists.</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be read from the
dialplan, and not directly from external protocols. It can, however, be
executed as a write operation (<literal>DB_DELETE(family, key)=ignored</literal>)</para>
</note>
</description>
<see-also>
<ref type="application">DBdel</ref>
<ref type="function">DB</ref>
<ref type="application">DBdeltree</ref>
</see-also>
</function>
<function name="CALLERID" language="en_US">
<synopsis>
Gets or sets Caller*ID data on the channel.
</synopsis>
<syntax>
<parameter name="datatype" required="true">
<para>The allowable datatypes are:</para>
<enumlist>
<enum name = "all" />
<enum name = "name" />
<enum name = "name-valid" />
<enum name = "name-charset" />
<enum name = "name-pres" />
<enum name = "num" />
<enum name = "num-valid" />
<enum name = "num-plan" />
<enum name = "num-pres" />
<enum name = "pres" />
<enum name = "subaddr" />
<enum name = "subaddr-valid" />
<enum name = "subaddr-type" />
<enum name = "subaddr-odd" />
<enum name = "tag" />
<enum name = "priv-all" />
<enum name = "priv-name" />
<enum name = "priv-name-valid" />
<enum name = "priv-name-charset" />
<enum name = "priv-name-pres" />
<enum name = "priv-num" />
<enum name = "priv-num-valid" />
<enum name = "priv-num-plan" />
<enum name = "priv-num-pres" />
<enum name = "priv-subaddr" />
<enum name = "priv-subaddr-valid" />
<enum name = "priv-subaddr-type" />
<enum name = "priv-subaddr-odd" />
<enum name = "priv-tag" />
<enum name = "ANI-all" />
<enum name = "ANI-name" />
<enum name = "ANI-name-valid" />
<enum name = "ANI-name-charset" />
<enum name = "ANI-name-pres" />
<enum name = "ANI-num" />
<enum name = "ANI-num-valid" />
<enum name = "ANI-num-plan" />
<enum name = "ANI-num-pres" />
<enum name = "ANI-tag" />
<enum name = "RDNIS" />
<enum name = "DNID" />
<enum name = "dnid-num-plan" />
<enum name = "dnid-subaddr" />
<enum name = "dnid-subaddr-valid" />
<enum name = "dnid-subaddr-type" />
<enum name = "dnid-subaddr-odd" />
</enumlist>
</parameter>
<parameter name="CID">
<para>Optional Caller*ID to parse instead of using the Caller*ID from the
channel. This parameter is only optional when reading the Caller*ID.</para>
</parameter>
</syntax>
<description>
<para>Gets or sets Caller*ID data on the channel. Uses channel callerid by
default or optional callerid, if specified.</para>
<para>The <replaceable>pres</replaceable> field gets/sets a combined value
for <replaceable>name-pres</replaceable> and
<replaceable>num-pres</replaceable>.</para>
<para>The allowable values for the <replaceable>name-charset</replaceable>
field are the following:</para>
<enumlist>
<enum name = "unknown">
<para>Unknown</para>
</enum>
<enum name = "iso8859-1">
<para>ISO8859-1</para>
</enum>
<enum name = "withdrawn">
<para>Withdrawn</para>
</enum>
<enum name = "iso8859-2">
<para>ISO8859-2</para>
</enum>
<enum name = "iso8859-3">
<para>ISO8859-3</para>
</enum>
<enum name = "iso8859-4">
<para>ISO8859-4</para>
</enum>
<enum name = "iso8859-5">
<para>ISO8859-5</para>
</enum>
<enum name = "iso8859-7">
<para>ISO8859-7</para>
</enum>
<enum name = "bmp">
<para>ISO10646 Bmp String</para>
</enum>
<enum name = "utf8">
<para>ISO10646 UTF-8 String</para>
</enum>
</enumlist>
</description>
</function>
<function name="CALLERPRES" language="en_US">
<synopsis>
Gets or sets Caller*ID presentation on the channel.
</synopsis>
<syntax />
<description>
<para>Gets or sets Caller*ID presentation on the channel.
This function is deprecated in favor of CALLERID(num-pres)
and CALLERID(name-pres) or CALLERID(pres) to get/set both
at once.
The following values are valid:</para>
<enumlist>
<enum name="allowed_not_screened">
<para>Presentation Allowed, Not Screened.</para>
</enum>
<enum name="allowed_passed_screen">
<para>Presentation Allowed, Passed Screen.</para>
</enum>
<enum name="allowed_failed_screen">
<para>Presentation Allowed, Failed Screen.</para>
</enum>
<enum name="allowed">
<para>Presentation Allowed, Network Number.</para>
</enum>
<enum name="prohib_not_screened">
<para>Presentation Prohibited, Not Screened.</para>
</enum>
<enum name="prohib_passed_screen">
<para>Presentation Prohibited, Passed Screen.</para>
</enum>
<enum name="prohib_failed_screen">
<para>Presentation Prohibited, Failed Screen.</para>
</enum>
<enum name="prohib">
<para>Presentation Prohibited, Network Number.</para>
</enum>
<enum name="unavailable">
<para>Number Unavailable.</para>
</enum>
</enumlist>
</description>
</function>
<function name="CONNECTEDLINE" language="en_US">
<synopsis>
Gets or sets Connected Line data on the channel.
</synopsis>
<syntax>
<parameter name="datatype" required="true">
<para>The allowable datatypes are:</para>
<enumlist>
<enum name = "all" />
<enum name = "name" />
<enum name = "name-valid" />
<enum name = "name-charset" />
<enum name = "name-pres" />
<enum name = "num" />
<enum name = "num-valid" />
<enum name = "num-plan" />
<enum name = "num-pres" />
<enum name = "pres" />
<enum name = "subaddr" />
<enum name = "subaddr-valid" />
<enum name = "subaddr-type" />
<enum name = "subaddr-odd" />
<enum name = "tag" />
<enum name = "priv-all" />
<enum name = "priv-name" />
<enum name = "priv-name-valid" />
<enum name = "priv-name-charset" />
<enum name = "priv-name-pres" />
<enum name = "priv-num" />
<enum name = "priv-num-valid" />
<enum name = "priv-num-plan" />
<enum name = "priv-num-pres" />
<enum name = "priv-subaddr" />
<enum name = "priv-subaddr-valid" />
<enum name = "priv-subaddr-type" />
<enum name = "priv-subaddr-odd" />
<enum name = "priv-tag" />
</enumlist>
</parameter>
<parameter name="i">
<para>If set, this will prevent the channel from sending out protocol
messages because of the value being set</para>
</parameter>
</syntax>
<description>
<para>Gets or sets Connected Line data on the channel.</para>
<para>The <replaceable>pres</replaceable> field gets/sets a combined value
for <replaceable>name-pres</replaceable> and
<replaceable>num-pres</replaceable>.</para>
<para>The allowable values for the <replaceable>name-charset</replaceable>
field are the following:</para>
<enumlist>
<enum name = "unknown">
<para>Unknown</para>
</enum>
<enum name = "iso8859-1">
<para>ISO8859-1</para>
</enum>
<enum name = "withdrawn">
<para>Withdrawn</para>
</enum>
<enum name = "iso8859-2">
<para>ISO8859-2</para>
</enum>
<enum name = "iso8859-3">
<para>ISO8859-3</para>
</enum>
<enum name = "iso8859-4">
<para>ISO8859-4</para>
</enum>
<enum name = "iso8859-5">
<para>ISO8859-5</para>
</enum>
<enum name = "iso8859-7">
<para>ISO8859-7</para>
</enum>
<enum name = "bmp">
<para>ISO10646 Bmp String</para>
</enum>
<enum name = "utf8">
<para>ISO10646 UTF-8 String</para>
</enum>
</enumlist>
</description>
</function>
<function name="REDIRECTING" language="en_US">
<synopsis>
Gets or sets Redirecting data on the channel.
</synopsis>
<syntax>
<parameter name="datatype" required="true">
<para>The allowable datatypes are:</para>
<enumlist>
<enum name = "orig-all" />
<enum name = "orig-name" />
<enum name = "orig-name-valid" />
<enum name = "orig-name-charset" />
<enum name = "orig-name-pres" />
<enum name = "orig-num" />
<enum name = "orig-num-valid" />
<enum name = "orig-num-plan" />
<enum name = "orig-num-pres" />
<enum name = "orig-pres" />
<enum name = "orig-subaddr" />
<enum name = "orig-subaddr-valid" />
<enum name = "orig-subaddr-type" />
<enum name = "orig-subaddr-odd" />
<enum name = "orig-tag" />
<enum name = "orig-reason" />
<enum name = "from-all" />
<enum name = "from-name" />
<enum name = "from-name-valid" />
<enum name = "from-name-charset" />
<enum name = "from-name-pres" />
<enum name = "from-num" />
<enum name = "from-num-valid" />
<enum name = "from-num-plan" />
<enum name = "from-num-pres" />
<enum name = "from-pres" />
<enum name = "from-subaddr" />
<enum name = "from-subaddr-valid" />
<enum name = "from-subaddr-type" />
<enum name = "from-subaddr-odd" />
<enum name = "from-tag" />
<enum name = "to-all" />
<enum name = "to-name" />
<enum name = "to-name-valid" />
<enum name = "to-name-charset" />
<enum name = "to-name-pres" />
<enum name = "to-num" />
<enum name = "to-num-valid" />
<enum name = "to-num-plan" />
<enum name = "to-num-pres" />
<enum name = "to-pres" />
<enum name = "to-subaddr" />
<enum name = "to-subaddr-valid" />
<enum name = "to-subaddr-type" />
<enum name = "to-subaddr-odd" />
<enum name = "to-tag" />
<enum name = "priv-orig-all" />
<enum name = "priv-orig-name" />
<enum name = "priv-orig-name-valid" />
<enum name = "priv-orig-name-charset" />
<enum name = "priv-orig-name-pres" />
<enum name = "priv-orig-num" />
<enum name = "priv-orig-num-valid" />
<enum name = "priv-orig-num-plan" />
<enum name = "priv-orig-num-pres" />
<enum name = "priv-orig-subaddr" />
<enum name = "priv-orig-subaddr-valid" />
<enum name = "priv-orig-subaddr-type" />
<enum name = "priv-orig-subaddr-odd" />
<enum name = "priv-orig-tag" />
<enum name = "priv-from-all" />
<enum name = "priv-from-name" />
<enum name = "priv-from-name-valid" />
<enum name = "priv-from-name-charset" />
<enum name = "priv-from-name-pres" />
<enum name = "priv-from-num" />
<enum name = "priv-from-num-valid" />
<enum name = "priv-from-num-plan" />
<enum name = "priv-from-num-pres" />
<enum name = "priv-from-subaddr" />
<enum name = "priv-from-subaddr-valid" />
<enum name = "priv-from-subaddr-type" />
<enum name = "priv-from-subaddr-odd" />
<enum name = "priv-from-tag" />
<enum name = "priv-to-all" />
<enum name = "priv-to-name" />
<enum name = "priv-to-name-valid" />
<enum name = "priv-to-name-charset" />
<enum name = "priv-to-name-pres" />
<enum name = "priv-to-num" />
<enum name = "priv-to-num-valid" />
<enum name = "priv-to-num-plan" />
<enum name = "priv-to-num-pres" />
<enum name = "priv-to-subaddr" />
<enum name = "priv-to-subaddr-valid" />
<enum name = "priv-to-subaddr-type" />
<enum name = "priv-to-subaddr-odd" />
<enum name = "priv-to-tag" />
<enum name = "reason" />
<enum name = "count" />
</enumlist>
</parameter>
<parameter name="i">
<para>If set, this will prevent the channel from sending out protocol
messages because of the value being set</para>
</parameter>
</syntax>
<description>
<para>Gets or sets Redirecting data on the channel.</para>
<para>The <replaceable>orig-pres</replaceable>,
<replaceable>from-pres</replaceable> and <replaceable>to-pres</replaceable>
fields get/set a combined value for the corresponding
<replaceable>...-name-pres</replaceable> and <replaceable>...-num-pres</replaceable>
fields.</para>
<para>The recognized values for the <replaceable>reason</replaceable>
and <replaceable>orig-reason</replaceable> fields are the following:</para>
<enumlist>
<enum name = "away">
<para>Callee is Away</para>
</enum>
<enum name = "cf_dte">
<para>Call Forwarding By The Called DTE</para>
</enum>
<enum name = "cfb">
<para>Call Forwarding Busy</para>
</enum>
<enum name = "cfnr">
<para>Call Forwarding No Reply</para>
</enum>
<enum name = "cfu">
<para>Call Forwarding Unconditional</para>
</enum>
<enum name = "deflection">
<para>Call Deflection</para>
</enum>
<enum name = "dnd">
<para>Do Not Disturb</para>
</enum>
<enum name = "follow_me">
<para>Follow Me</para>
</enum>
<enum name = "out_of_order">
<para>Called DTE Out-Of-Order</para>
</enum>
<enum name = "send_to_vm">
<para>Send the call to voicemail</para>
</enum>
<enum name = "time_of_day">
<para>Time of Day</para>
</enum>
<enum name = "unavailable">
<para>Callee is Unavailable</para>
</enum>
<enum name = "unknown">
<para>Unknown</para>
</enum>
</enumlist>
<note>
<para>You can set a user defined reason string that SIP can
send/receive instead. The user defined reason string my need to be
quoted depending upon SIP or the peer's requirements. These strings
are treated as unknown by the non-SIP channel drivers.</para>
</note>
<para>The allowable values for the <replaceable>xxx-name-charset</replaceable>
field are the following:</para>
<enumlist>
<enum name = "unknown">
<para>Unknown</para>
</enum>
<enum name = "iso8859-1">
<para>ISO8859-1</para>
</enum>
<enum name = "withdrawn">
<para>Withdrawn</para>
</enum>
<enum name = "iso8859-2">
<para>ISO8859-2</para>
</enum>
<enum name = "iso8859-3">
<para>ISO8859-3</para>
</enum>
<enum name = "iso8859-4">
<para>ISO8859-4</para>
</enum>
<enum name = "iso8859-5">
<para>ISO8859-5</para>
</enum>
<enum name = "iso8859-7">
<para>ISO8859-7</para>
</enum>
<enum name = "bmp">
<para>ISO10646 Bmp String</para>
</enum>
<enum name = "utf8">
<para>ISO10646 UTF-8 String</para>
</enum>
</enumlist>
</description>
</function>
<function name="SHELL" language="en_US">
<synopsis>
Executes a command using the system shell and captures its output.
</synopsis>
<syntax>
<parameter name="command" required="true">
<para>The command that the shell should execute.</para>
<warning>
<para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</parameter>
</syntax>
<description>
<para>Collects the output generated by a command executed by the system shell</para>
<para>Example: <literal>Set(foo=${SHELL(echo bar)})</literal></para>
<note>
<para>The command supplied to this function will be executed by the
system's shell, typically specified in the SHELL environment variable. There
are many different system shells available with somewhat different behaviors,
so the output generated by this function may vary between platforms.</para>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
</function>
<function name="CALLCOMPLETION" language="en_US">
<synopsis>
Get or set a call completion configuration parameter for a channel.
</synopsis>
<syntax>
<parameter name="option" required="true">
<para>The allowable options are:</para>
<enumlist>
<enum name="cc_agent_policy" />
<enum name="cc_monitor_policy" />
<enum name="cc_offer_timer" />
<enum name="ccnr_available_timer" />
<enum name="ccbs_available_timer" />
<enum name="cc_recall_timer" />
<enum name="cc_max_agents" />
<enum name="cc_max_monitors" />
<enum name="cc_callback_macro" />
<enum name="cc_agent_dialstring" />
</enumlist>
</parameter>
</syntax>
<description>
<para>The CALLCOMPLETION function can be used to get or set a call
completion configuration parameter for a channel. Note that setting
a configuration parameter will only change the parameter for the
duration of the call.
For more information see <filename>doc/AST.pdf</filename>.
For more information on call completion parameters, see <filename>configs/ccss.conf.sample</filename>.</para>
</description>
</function>
<function name="CDR" language="en_US">
<synopsis>
Gets or sets a CDR variable.
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>CDR field name:</para>
<enumlist>
<enum name="clid">
<para>Caller ID.</para>
</enum>
<enum name="lastdata">
<para>Last application arguments.</para>
</enum>
<enum name="disposition">
<para>The final state of the CDR.</para>
<enumlist>
<enum name="0">
<para>
<literal>NO ANSWER</literal>
</para>
</enum>
<enum name="1">
<para>
<literal>NO ANSWER</literal> (NULL record)</para>
</enum>
<enum name="2">
<para>
<literal>FAILED</literal>
</para>
</enum>
<enum name="4">
<para>
<literal>BUSY</literal>
</para>
</enum>
<enum name="8">
<para>
<literal>ANSWERED</literal>
</para>
</enum>
<enum name="16">
<para>
<literal>CONGESTION</literal>
</para>
</enum>
</enumlist>
</enum>
<enum name="src">
<para>Source.</para>
</enum>
<enum name="start">
<para>Time the call started.</para>
</enum>
<enum name="amaflags">
<para>R/W the Automatic Message Accounting (AMA) flags on the channel.
When read from a channel, the integer value will always be returned.
When written to a channel, both the string format or integer value
is accepted.</para>
<enumlist>
<enum name="1">
<para>
<literal>OMIT</literal>
</para>
</enum>
<enum name="2">
<para>
<literal>BILLING</literal>
</para>
</enum>
<enum name="3">
<para>
<literal>DOCUMENTATION</literal>
</para>
</enum>
</enumlist>
<warning>
<para>Accessing this setting is deprecated in CDR. Please use the CHANNEL function instead.</para>
</warning>
</enum>
<enum name="dst">
<para>Destination.</para>
</enum>
<enum name="answer">
<para>Time the call was answered.</para>
</enum>
<enum name="accountcode">
<para>The channel's account code.</para>
<warning>
<para>Accessing this setting is deprecated in CDR. Please use the CHANNEL function instead.</para>
</warning>
</enum>
<enum name="dcontext">
<para>Destination context.</para>
</enum>
<enum name="end">
<para>Time the call ended.</para>
</enum>
<enum name="uniqueid">
<para>The channel's unique id.</para>
</enum>
<enum name="dstchannel">
<para>Destination channel.</para>
</enum>
<enum name="duration">
<para>Duration of the call.</para>
</enum>
<enum name="userfield">
<para>The channel's user specified field.</para>
</enum>
<enum name="lastapp">
<para>Last application.</para>
</enum>
<enum name="billsec">
<para>Duration of the call once it was answered.</para>
</enum>
<enum name="channel">
<para>Channel name.</para>
</enum>
<enum name="sequence">
<para>CDR sequence number.</para>
</enum>
</enumlist>
</parameter>
<parameter name="options" required="false">
<optionlist>
<option name="f">
<para>Returns billsec or duration fields as floating point values.</para>
</option>
<option name="u">
<para>Retrieves the raw, unprocessed value.</para>
<para>For example, 'start', 'answer', and 'end' will be retrieved as epoch
values, when the <literal>u</literal> option is passed, but formatted as YYYY-MM-DD HH:MM:SS
otherwise. Similarly, disposition and amaflags will return their raw
integral values.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>All of the CDR field names are read-only, except for <literal>accountcode</literal>,
<literal>userfield</literal>, and <literal>amaflags</literal>. You may, however, supply
a name not on the above list, and create your own variable, whose value can be changed
with this function, and this variable will be stored on the CDR.</para>
<note>
<para>CDRs can only be modified before the bridge between two channels is
torn down. For example, CDRs may not be modified after the <literal>Dial</literal>
application has returned.</para>
</note>
<para>Example: exten => 1,1,Set(CDR(userfield)=test)</para>
</description>
</function>
<function name="CDR_PROP" language="en_US">
<synopsis>
Set a property on a channel's CDR.
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The property to set on the CDR.</para>
<enumlist>
<enum name="party_a">
<para>Set this channel as the preferred Party A when
channels are associated together.</para>
<para>Write-Only</para>
</enum>
<enum name="disable">
<para>Setting to 1 will disable CDRs for this channel.
Setting to 0 will enable CDRs for this channel.</para>
<para>Write-Only</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function sets a property on a channel's CDR. Properties
alter the behavior of how the CDR operates for that channel.</para>
</description>
</function>
<function name="BLACKLIST" language="en_US">
<synopsis>
Check if the callerid is on the blacklist.
</synopsis>
<syntax />
<description>
<para>Uses astdb to check if the Caller*ID is in family <literal>blacklist</literal>.
Returns <literal>1</literal> or <literal>0</literal>.</para>
</description>
<see-also>
<ref type="function">DB</ref>
</see-also>
</function>
<function name="BASE64_ENCODE" language="en_US">
<synopsis>
Encode a string in base64.
</synopsis>
<syntax>
<parameter name="string" required="true">
<para>Input string</para>
</parameter>
</syntax>
<description>
<para>Returns the base64 string.</para>
</description>
<see-also>
<ref type="function">BASE64_DECODE</ref>
<ref type="function">AES_DECRYPT</ref>
<ref type="function">AES_ENCRYPT</ref>
</see-also>
</function>
<function name="BASE64_DECODE" language="en_US">
<synopsis>
Decode a base64 string.
</synopsis>
<syntax>
<parameter name="string" required="true">
<para>Input string.</para>
</parameter>
</syntax>
<description>
<para>Returns the plain text string.</para>
</description>
<see-also>
<ref type="function">BASE64_ENCODE</ref>
<ref type="function">AES_DECRYPT</ref>
<ref type="function">AES_ENCRYPT</ref>
</see-also>
</function>
<function name="SYSINFO" language="en_US">
<synopsis>
Returns system information specified by parameter.
</synopsis>
<syntax>
<parameter name="parameter" required="true">
<enumlist>
<enum name="loadavg">
<para>System load average from past minute.</para>
</enum>
<enum name="numcalls">
<para>Number of active calls currently in progress.</para>
</enum>
<enum name="uptime">
<para>System uptime in hours.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="totalram">
<para>Total usable main memory size in KiB.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="freeram">
<para>Available memory size in KiB.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="bufferram">
<para>Memory used by buffers in KiB.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="totalswap">
<para>Total swap space still available in KiB.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="freeswap">
<para>Free swap space still available in KiB.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
<enum name="numprocs">
<para>Number of current processes.</para>
<note>
<para>This parameter is dependant upon operating system.</para>
</note>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Returns information from a given parameter.</para>
</description>
</function>
<function name="GLOBAL" language="en_US">
<synopsis>
Gets or sets the global variable specified.
</synopsis>
<syntax>
<parameter name="varname" required="true">
<para>Global variable name</para>
</parameter>
</syntax>
<description>
<para>Set or get the value of a global variable specified in <replaceable>varname</replaceable></para>
</description>
</function>
<function name="SHARED" language="en_US">
<synopsis>
Gets or sets the shared variable specified.
</synopsis>
<syntax>
<parameter name="varname" required="true">
<para>Variable name</para>
</parameter>
<parameter name="channel">
<para>If not specified will default to current channel. It is the complete
channel name: <literal>SIP/12-abcd1234</literal> or the prefix only <literal>SIP/12</literal>.</para>
</parameter>
</syntax>
<description>
<para>Implements a shared variable area, in which you may share variables between
channels.</para>
<para>The variables used in this space are separate from the general namespace of
the channel and thus <variable>SHARED(foo)</variable> and <variable>foo</variable>
represent two completely different variables, despite sharing the same name.</para>
<para>Finally, realize that there is an inherent race between channels operating
at the same time, fiddling with each others' internal variables, which is why
this special variable namespace exists; it is to remind you that variables in
the SHARED namespace may change at any time, without warning. You should
therefore take special care to ensure that when using the SHARED namespace,
you retrieve the variable and store it in a regular channel variable before
using it in a set of calculations (or you might be surprised by the result).</para>
</description>
</function>
<managerEvent language="en_US" name="VarSet">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>Raised when a variable is shared between channels.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Variable">
<para>The SHARED variable being set.</para>
<note>
<para>The variable name will always be enclosed with
<literal>SHARED()</literal>
</para>
</note>
</parameter>
<parameter name="Value">
<para>The new value of the variable.</para>
</parameter>
</syntax>
<see-also>
<ref type="function">SHARED</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<function name="URIENCODE" language="en_US">
<synopsis>
Encodes a string to URI-safe encoding according to RFC 2396.
</synopsis>
<syntax>
<parameter name="data" required="true">
<para>Input string to be encoded.</para>
</parameter>
</syntax>
<description>
<para>Returns the encoded string defined in <replaceable>data</replaceable>.</para>
</description>
</function>
<function name="URIDECODE" language="en_US">
<synopsis>
Decodes a URI-encoded string according to RFC 2396.
</synopsis>
<syntax>
<parameter name="data" required="true">
<para>Input string to be decoded.</para>
</parameter>
</syntax>
<description>
<para>Returns the decoded URI-encoded <replaceable>data</replaceable> string.</para>
</description>
</function>
<function name="PJSIP_CONTACT" language="en_US">
<synopsis>
Get information about a PJSIP contact
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the contact to query.</para>
</parameter>
<parameter name="field" required="true">
<para>The configuration option for the contact to query for.
Supported options are those fields on the
<replaceable>contact</replaceable> object.</para>
<enumlist>
<configOptionToEnum>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='contact']/configOption)"/>
</configOptionToEnum>
</enumlist>
</parameter>
</syntax>
</function>
<function name="SHA1" language="en_US">
<synopsis>
Computes a SHA1 digest.
</synopsis>
<syntax>
<parameter name="data" required="true">
<para>Input string</para>
</parameter>
</syntax>
<description>
<para>Generate a SHA1 digest via the SHA1 algorythm.</para>
<para>Example: Set(sha1hash=${SHA1(junky)})</para>
<para>Sets the asterisk variable sha1hash to the string <literal>60fa5675b9303eb62f99a9cd47f9f5837d18f9a0</literal>
which is known as his hash</para>
</description>
</function>
<function name="DIALPLAN_EXISTS" language="en_US">
<synopsis>
Checks the existence of a dialplan target.
</synopsis>
<syntax>
<parameter name="context" required="true" />
<parameter name="extension" />
<parameter name="priority" />
</syntax>
<description>
<para>This function returns <literal>1</literal> if the target exits. Otherwise, it returns <literal>0</literal>.</para>
</description>
</function>
<function name="VALID_EXTEN" language="en_US">
<synopsis>
Determine whether an extension exists or not.
</synopsis>
<syntax>
<parameter name="context">
<para>Defaults to the current context</para>
</parameter>
<parameter name="extension" required="true" />
<parameter name="priority">
<para>Priority defaults to <literal>1</literal>.</para>
</parameter>
</syntax>
<description>
<para>Returns a true value if the indicated <replaceable>context</replaceable>,
<replaceable>extension</replaceable>, and <replaceable>priority</replaceable> exist.</para>
<warning>
<para>This function has been deprecated in favor of the <literal>DIALPLAN_EXISTS()</literal> function</para>
</warning>
</description>
</function>
<function name="HOLD_INTERCEPT" language="en_US">
<synopsis>
Intercepts hold frames on a channel and raises an event instead of passing the frame on
</synopsis>
<syntax>
<parameter name="action" required="true">
<optionlist>
<option name="remove">
<para>W/O. Removes the hold interception function.</para>
</option>
<option name="set">
<para>W/O. Enable hold interception on the channel. When
enabled, the channel will intercept any hold action that
is signalled from the device, and instead simply raise an
event (AMI/ARI) indicating that the channel wanted to put other
parties on hold.</para>
</option>
</optionlist>
</parameter>
</syntax>
</function>
<function name="RAND" language="en_US">
<synopsis>
Choose a random number in a range.
</synopsis>
<syntax>
<parameter name="min" />
<parameter name="max" />
</syntax>
<description>
<para>Choose a random number between <replaceable>min</replaceable> and <replaceable>max</replaceable>.
<replaceable>min</replaceable> defaults to <literal>0</literal>, if not specified, while <replaceable>max</replaceable> defaults
to <literal>RAND_MAX</literal> (2147483647 on many systems).</para>
<para>Example: Set(junky=${RAND(1,8)});
Sets junky to a random number between 1 and 8, inclusive.</para>
</description>
</function>
<function name="VMCOUNT" language="en_US">
<synopsis>
Count the voicemails in a specified mailbox.
</synopsis>
<syntax>
<parameter name="vmbox" required="true" />
<parameter name="folder" required="false">
<para>If not specified, defaults to <literal>INBOX</literal></para>
</parameter>
</syntax>
<description>
<para>Count the number of voicemails in a specified mailbox, you could also specify
the mailbox <replaceable>folder</replaceable>.</para>
<para>Example: <literal>exten => s,1,Set(foo=${VMCOUNT(125@default)})</literal></para>
</description>
</function>
<function name="LOCK" language="en_US">
<synopsis>
Attempt to obtain a named mutex.
</synopsis>
<syntax>
<parameter name="lockname" required="true" />
</syntax>
<description>
<para>Attempts to grab a named lock exclusively, and prevents other channels from
obtaining the same lock. LOCK will wait for the lock to become available.
Returns <literal>1</literal> if the lock was obtained or <literal>0</literal> on error.</para>
<note>
<para>To avoid the possibility of a deadlock, LOCK will only attempt to
obtain the lock for 3 seconds if the channel already has another lock.</para>
</note>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
</function>
<function name="TRYLOCK" language="en_US">
<synopsis>
Attempt to obtain a named mutex.
</synopsis>
<syntax>
<parameter name="lockname" required="true" />
</syntax>
<description>
<para>Attempts to grab a named lock exclusively, and prevents other channels
from obtaining the same lock. Returns <literal>1</literal> if the lock was
available or <literal>0</literal> otherwise.</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
</function>
<function name="UNLOCK" language="en_US">
<synopsis>
Unlocks a named mutex.
</synopsis>
<syntax>
<parameter name="lockname" required="true" />
</syntax>
<description>
<para>Unlocks a previously locked mutex. Returns <literal>1</literal> if the channel
had a lock or <literal>0</literal> otherwise.</para>
<note>
<para>It is generally unnecessary to unlock in a hangup routine, as any locks
held are automatically freed when the channel is destroyed.</para>
</note>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
</function>
<function name="DEVICE_STATE" language="en_US">
<synopsis>
Get or Set a device state.
</synopsis>
<syntax>
<parameter name="device" required="true" />
</syntax>
<description>
<para>The DEVICE_STATE function can be used to retrieve the device state from any
device state provider. For example:</para>
<para>NoOp(SIP/mypeer has state ${DEVICE_STATE(SIP/mypeer)})</para>
<para>NoOp(Conference number 1234 has state ${DEVICE_STATE(MeetMe:1234)})</para>
<para>The DEVICE_STATE function can also be used to set custom device state from
the dialplan. The <literal>Custom:</literal> prefix must be used. For example:</para>
<para>Set(DEVICE_STATE(Custom:lamp1)=BUSY)</para>
<para>Set(DEVICE_STATE(Custom:lamp2)=NOT_INUSE)</para>
<para>You can subscribe to the status of a custom device state using a hint in
the dialplan:</para>
<para>exten => 1234,hint,Custom:lamp1</para>
<para>The possible values for both uses of this function are:</para>
<para>UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING |
RINGINUSE | ONHOLD</para>
</description>
</function>
<function name="HINT" language="en_US">
<synopsis>
Get the devices set for a dialplan hint.
</synopsis>
<syntax>
<parameter name="extension" required="true" argsep="@">
<argument name="extension" required="true" />
<argument name="context" />
</parameter>
<parameter name="options">
<optionlist>
<option name="n">
<para>Retrieve name on the hint instead of list of devices.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The HINT function can be used to retrieve the list of devices that are
mapped to a dialplan hint. For example:</para>
<para>NoOp(Hint for Extension 1234 is ${HINT(1234)})</para>
</description>
</function>
<function name="PJSIP_AOR" language="en_US">
<synopsis>
Get information about a PJSIP AOR
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the AOR to query.</para>
</parameter>
<parameter name="field" required="true">
<para>The configuration option for the AOR to query for.
Supported options are those fields on the
<replaceable>aor</replaceable> object in
<filename>pjsip.conf</filename>.</para>
<enumlist>
<configOptionToEnum>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption)"/>
</configOptionToEnum>
</enumlist>
</parameter>
</syntax>
</function>
<function name="AST_SORCERY" language="en_US">
<synopsis>
Get a field from a sorcery object
</synopsis>
<syntax>
<parameter name="module_name" required="true">
<para>The name of the module owning the sorcery instance.</para>
</parameter>
<parameter name="object_type" required="true">
<para>The type of object to query.</para>
</parameter>
<parameter name="object_id" required="true">
<para>The id of the object to query.</para>
</parameter>
<parameter name="field_name" required="true">
<para>The name of the field.</para>
</parameter>
<parameter name="retrieval_method" required="false">
<para>Fields that have multiple occurrences may be retrieved in two ways.</para>
<enumlist>
<enum name="concat">
<para>Returns all matching fields concatenated
in a single string separated by <replaceable>separator</replaceable>
which defaults to <literal>,</literal>.</para>
</enum>
<enum name="single">
<para>Returns the nth occurrence of the field
as specified by <replaceable>occurrence_number</replaceable> which defaults to <literal>1</literal>.
</para>
</enum>
</enumlist>
<para>The default is <literal>concat</literal> with separator <literal>,</literal>.</para>
</parameter>
<parameter name="retrieval_details" required="false">
<para>Specifies either the separator for <literal>concat</literal>
or the occurrence number for <literal>single</literal>.</para>
</parameter>
</syntax>
</function>
<function name="AES_ENCRYPT" language="en_US">
<synopsis>
Encrypt a string with AES given a 16 character key.
</synopsis>
<syntax>
<parameter name="key" required="true">
<para>AES Key</para>
</parameter>
<parameter name="string" required="true">
<para>Input string</para>
</parameter>
</syntax>
<description>
<para>Returns an AES encrypted string encoded in base64.</para>
</description>
<see-also>
<ref type="function">AES_DECRYPT</ref>
<ref type="function">BASE64_ENCODE</ref>
<ref type="function">BASE64_DECODE</ref>
</see-also>
</function>
<function name="AES_DECRYPT" language="en_US">
<synopsis>
Decrypt a string encoded in base64 with AES given a 16 character key.
</synopsis>
<syntax>
<parameter name="key" required="true">
<para>AES Key</para>
</parameter>
<parameter name="string" required="true">
<para>Input string.</para>
</parameter>
</syntax>
<description>
<para>Returns the plain text string.</para>
</description>
<see-also>
<ref type="function">AES_ENCRYPT</ref>
<ref type="function">BASE64_ENCODE</ref>
<ref type="function">BASE64_DECODE</ref>
</see-also>
</function>
<function name="FIELDQTY" language="en_US">
<synopsis>
Count the fields with an arbitrary delimiter
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delim" required="true" />
</syntax>
<description>
<para>The delimiter may be specified as a special or extended ASCII character, by encoding it. The characters
<literal>\n</literal>, <literal>\r</literal>, and <literal>\t</literal> are all recognized as the newline,
carriage return, and tab characters, respectively. Also, octal and hexadecimal specifications are recognized
by the patterns <literal>\0nnn</literal> and <literal>\xHH</literal>, respectively. For example, if you wanted
to encode a comma as the delimiter, you could use either <literal>\054</literal> or <literal>\x2C</literal>.</para>
<para>Example: If ${example} contains <literal>ex-amp-le</literal>, then ${FIELDQTY(example,-)} returns 3.</para>
</description>
</function>
<function name="FIELDNUM" language="en_US">
<synopsis>
Return the 1-based offset of a field in a list
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delim" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Search the variable named <replaceable>varname</replaceable> for the string <replaceable>value</replaceable>
delimited by <replaceable>delim</replaceable> and return a 1-based offset as to its location. If not found
or an error occured, return <literal>0</literal>.</para>
<para>The delimiter may be specified as a special or extended ASCII character, by encoding it. The characters
<literal>\n</literal>, <literal>\r</literal>, and <literal>\t</literal> are all recognized as the newline,
carriage return, and tab characters, respectively. Also, octal and hexadecimal specifications are recognized
by the patterns <literal>\0nnn</literal> and <literal>\xHH</literal>, respectively. For example, if you wanted
to encode a comma as the delimiter, you could use either <literal>\054</literal> or <literal>\x2C</literal>.</para>
<para>Example: If ${example} contains <literal>ex-amp-le</literal>, then ${FIELDNUM(example,-,amp)} returns 2.</para>
</description>
</function>
<function name="LISTFILTER" language="en_US">
<synopsis>Remove an item from a list, by name.</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delim" required="true" default="," />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Remove <replaceable>value</replaceable> from the list contained in the <replaceable>varname</replaceable>
variable, where the list delimiter is specified by the <replaceable>delim</replaceable> parameter. This is
very useful for removing a single channel name from a list of channels, for example.</para>
</description>
</function>
<function name="FILTER" language="en_US">
<synopsis>
Filter the string to include only the allowed characters
</synopsis>
<syntax>
<parameter name="allowed-chars" required="true" />
<parameter name="string" required="true" />
</syntax>
<description>
<para>Permits all characters listed in <replaceable>allowed-chars</replaceable>,
filtering all others outs. In addition to literally listing the characters,
you may also use ranges of characters (delimited by a <literal>-</literal>
</para>
<para>Hexadecimal characters started with a <literal>\x</literal>(i.e. \x20)</para>
<para>Octal characters started with a <literal>\0</literal> (i.e. \040)</para>
<para>Also <literal>\t</literal>,<literal>\n</literal> and <literal>\r</literal> are recognized.</para>
<note>
<para>If you want the <literal>-</literal> character it needs to be prefixed with a
<literal>\</literal>
</para>
</note>
</description>
</function>
<function name="REPLACE" language="en_US">
<synopsis>
Replace a set of characters in a given string with another character.
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="find-chars" required="true" />
<parameter name="replace-char" required="false" />
</syntax>
<description>
<para>Iterates through a string replacing all the <replaceable>find-chars</replaceable> with
<replaceable>replace-char</replaceable>. <replaceable>replace-char</replaceable> may be either
empty or contain one character. If empty, all <replaceable>find-chars</replaceable> will be
deleted from the output.</para>
<note>
<para>The replacement only occurs in the output. The original variable is not
altered.</para>
</note>
</description>
</function>
<function name="STRREPLACE" language="en_US">
<synopsis>
Replace instances of a substring within a string with another string.
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="find-string" required="true" />
<parameter name="replace-string" required="false" />
<parameter name="max-replacements" required="false" />
</syntax>
<description>
<para>Searches for all instances of the <replaceable>find-string</replaceable> in provided variable and
replaces them with <replaceable>replace-string</replaceable>. If <replaceable>replace-string</replaceable>
is an empty string, this will effecively delete that substring. If <replaceable>max-replacements</replaceable>
is specified, this function will stop after performing replacements <replaceable>max-replacements</replaceable> times.</para>
<note>
<para>The replacement only occurs in the output. The original variable is not altered.</para>
</note>
</description>
</function>
<function name="PASSTHRU" language="en_US">
<synopsis>
Pass the given argument back as a value.
</synopsis>
<syntax>
<parameter name="string" required="false" />
</syntax>
<description>
<para>Literally returns the given <replaceable>string</replaceable>. The intent is to permit
other dialplan functions which take a variable name as an argument to be able to take a literal
string, instead.</para>
<note>
<para>The functions which take a variable name need to be passed var and not
${var}. Similarly, use PASSTHRU() and not ${PASSTHRU()}.</para>
</note>
<para>Example: ${CHANNEL} contains SIP/321-1</para>
<para> ${CUT(PASSTHRU(${CUT(CHANNEL,-,1)}),/,2)}) will return 321</para>
</description>
</function>
<function name="REGEX" language="en_US">
<synopsis>
Check string against a regular expression.
</synopsis>
<syntax argsep=" ">
<parameter name="&quot;regular expression&quot;" required="true" />
<parameter name="string" required="true" />
</syntax>
<description>
<para>Return <literal>1</literal> on regular expression match or <literal>0</literal> otherwise</para>
<para>Please note that the space following the double quotes separating the
regex from the data is optional and if present, is skipped. If a space is
desired at the beginning of the data, then put two spaces there; the second
will not be skipped.</para>
</description>
</function>
<application name="ClearHash" language="en_US">
<synopsis>
Clear the keys from a specified hashname.
</synopsis>
<syntax>
<parameter name="hashname" required="true" />
</syntax>
<description>
<para>Clears all keys out of the specified <replaceable>hashname</replaceable>.</para>
</description>
</application>
<function name="HASH" language="en_US">
<synopsis>
Implementation of a dialplan associative array
</synopsis>
<syntax>
<parameter name="hashname" required="true" />
<parameter name="hashkey" />
</syntax>
<description>
<para>In two arguments mode, gets and sets values to corresponding keys within
a named associative array. The single-argument mode will only work when assigned
to from a function defined by func_odbc</para>
</description>
</function>
<function name="HASHKEYS" language="en_US">
<synopsis>
Retrieve the keys of the HASH() function.
</synopsis>
<syntax>
<parameter name="hashname" required="true" />
</syntax>
<description>
<para>Returns a comma-delimited list of the current keys of the associative array
defined by the HASH() function. Note that if you iterate over the keys of
the result, adding keys during iteration will cause the result of the HASHKEYS()
function to change.</para>
</description>
</function>
<function name="KEYPADHASH" language="en_US">
<synopsis>
Hash the letters in string into equivalent keypad numbers.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${KEYPADHASH(Les)} returns "537"</para>
</description>
</function>
<function name="ARRAY" language="en_US">
<synopsis>
Allows setting multiple variables at once.
</synopsis>
<syntax>
<parameter name="var1" required="true" />
<parameter name="var2" required="false" multiple="true" />
<parameter name="varN" required="false" />
</syntax>
<description>
<para>The comma-delimited list passed as a value to which the function is set will
be interpreted as a set of values to which the comma-delimited list of
variable names in the argument should be set.</para>
<para>Example: Set(ARRAY(var1,var2)=1,2) will set var1 to 1 and var2 to 2</para>
</description>
</function>
<function name="STRPTIME" language="en_US">
<synopsis>
Returns the epoch of the arbitrary date/time string structured as described by the format.
</synopsis>
<syntax>
<parameter name="datetime" required="true" />
<parameter name="timezone" required="true" />
<parameter name="format" required="true" />
</syntax>
<description>
<para>This is useful for converting a date into <literal>EPOCH</literal> time,
possibly to pass to an application like SayUnixTime or to calculate the difference
between the two date strings</para>
<para>Example: ${STRPTIME(2006-03-01 07:30:35,America/Chicago,%Y-%m-%d %H:%M:%S)} returns 1141219835</para>
</description>
</function>
<function name="STRFTIME" language="en_US">
<synopsis>
Returns the current date/time in the specified format.
</synopsis>
<syntax>
<parameter name="epoch" />
<parameter name="timezone" />
<parameter name="format" />
</syntax>
<description>
<para>STRFTIME supports all of the same formats as the underlying C function
<emphasis>strftime(3)</emphasis>.
It also supports the following format: <literal>%[n]q</literal> - fractions of a second,
with leading zeros.</para>
<para>Example: <literal>%3q</literal> will give milliseconds and <literal>%1q</literal>
will give tenths of a second. The default is set at milliseconds (n=3).
The common case is to use it in combination with %S, as in <literal>%S.%3q</literal>.</para>
</description>
<see-also>
<ref type="manpage">strftime(3)</ref>
</see-also>
</function>
<function name="EVAL" language="en_US">
<synopsis>
Evaluate stored variables
</synopsis>
<syntax>
<parameter name="variable" required="true" />
</syntax>
<description>
<para>Using EVAL basically causes a string to be evaluated twice.
When a variable or expression is in the dialplan, it will be
evaluated at runtime. However, if the results of the evaluation
is in fact another variable or expression, using EVAL will have it
evaluated a second time.</para>
<para>Example: If the <variable>MYVAR</variable> contains
<variable>OTHERVAR</variable>, then the result of ${EVAL(
<variable>MYVAR</variable>)} in the dialplan will be the
contents of <variable>OTHERVAR</variable>. Normally just
putting <variable>MYVAR</variable> in the dialplan the result
would be <variable>OTHERVAR</variable>.</para>
</description>
</function>
<function name="TOUPPER" language="en_US">
<synopsis>
Convert string to all uppercase letters.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${TOUPPER(Example)} returns "EXAMPLE"</para>
</description>
</function>
<function name="TOLOWER" language="en_US">
<synopsis>
Convert string to all lowercase letters.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${TOLOWER(Example)} returns "example"</para>
</description>
</function>
<function name="LEN" language="en_US">
<synopsis>
Return the length of the string given.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${LEN(example)} returns 7</para>
</description>
</function>
<function name="QUOTE" language="en_US">
<synopsis>
Quotes a given string, escaping embedded quotes as necessary
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${QUOTE(ab"c"de)} will return ""ab\"c\"de""</para>
</description>
</function>
<function name="CSV_QUOTE" language="en_US">
<synopsis>
Quotes a given string for use in a CSV file, escaping embedded quotes as necessary
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>Example: ${CSV_QUOTE("a,b" 123)} will return """a,b"" 123"</para>
</description>
</function>
<function name="SHIFT" language="en_US">
<synopsis>
Removes and returns the first item off of a variable containing delimited text
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delimiter" required="false" default="," />
</syntax>
<description>
<para>Example:</para>
<para>exten => s,1,Set(array=one,two,three)</para>
<para>exten => s,n,While($["${SET(var=${SHIFT(array)})}" != ""])</para>
<para>exten => s,n,NoOp(var is ${var})</para>
<para>exten => s,n,EndWhile</para>
<para>This would iterate over each value in array, left to right, and
would result in NoOp(var is one), NoOp(var is two), and
NoOp(var is three) being executed.
</para>
</description>
</function>
<function name="POP" language="en_US">
<synopsis>
Removes and returns the last item off of a variable containing delimited text
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delimiter" required="false" default="," />
</syntax>
<description>
<para>Example:</para>
<para>exten => s,1,Set(array=one,two,three)</para>
<para>exten => s,n,While($["${SET(var=${POP(array)})}" != ""])</para>
<para>exten => s,n,NoOp(var is ${var})</para>
<para>exten => s,n,EndWhile</para>
<para>This would iterate over each value in array, right to left, and
would result in NoOp(var is three), NoOp(var is two), and
NoOp(var is one) being executed.
</para>
</description>
</function>
<function name="PUSH" language="en_US">
<synopsis>
Appends one or more values to the end of a variable containing delimited text
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delimiter" required="false" default="," />
</syntax>
<description>
<para>Example: Set(PUSH(array)=one,two,three) would append one,
two, and three to the end of the values stored in the variable
"array".
</para>
</description>
</function>
<function name="UNSHIFT" language="en_US">
<synopsis>
Inserts one or more values to the beginning of a variable containing delimited text
</synopsis>
<syntax>
<parameter name="varname" required="true" />
<parameter name="delimiter" required="false" default="," />
</syntax>
<description>
<para>Example: Set(UNSHIFT(array)=one,two,three) would insert one,
two, and three before the values stored in the variable
"array".
</para>
</description>
</function>
<function name="MATH" language="en_US">
<synopsis>
Performs Mathematical Functions.
</synopsis>
<syntax>
<parameter name="expression" required="true">
<para>Is of the form:
<replaceable>number1</replaceable>
<replaceable>op</replaceable>
<replaceable>number2</replaceable>
where the possible values for <replaceable>op</replaceable>
are:</para>
<para>+,-,/,*,%,&lt;&lt;,&gt;&gt;,^,AND,OR,XOR,&lt;,&gt;,&lt;=,&gt;=,== (and behave as their C equivalents)</para>
</parameter>
<parameter name="type">
<para>Wanted type of result:</para>
<para>f, float - float(default)</para>
<para>i, int - integer</para>
<para>h, hex - hex</para>
<para>c, char - char</para>
</parameter>
</syntax>
<description>
<para>Performs mathematical functions based on two parameters and an operator. The returned
value type is <replaceable>type</replaceable>
</para>
<para>Example: Set(i=${MATH(123%16,int)}) - sets var i=11</para>
</description>
</function>
<function name="INC" language="en_US">
<synopsis>
Increments the value of a variable, while returning the updated value to the dialplan
</synopsis>
<syntax>
<parameter name="variable" required="true">
<para>
The variable name to be manipulated, without the braces.
</para>
</parameter>
</syntax>
<description>
<para>Increments the value of a variable, while returning the updated value to the dialplan</para>
<para>Example: INC(MyVAR) - Increments MyVar</para>
<para>Note: INC(${MyVAR}) - Is wrong, as INC expects the variable name, not its value</para>
</description>
</function>
<function name="DEC" language="en_US">
<synopsis>
Decrements the value of a variable, while returning the updated value to the dialplan
</synopsis>
<syntax>
<parameter name="variable" required="true">
<para>
The variable name to be manipulated, without the braces.
</para>
</parameter>
</syntax>
<description>
<para>Decrements the value of a variable, while returning the updated value to the dialplan</para>
<para>Example: DEC(MyVAR) - Decrements MyVar</para>
<para>Note: DEC(${MyVAR}) - Is wrong, as DEC expects the variable name, not its value</para>
</description>
</function>
<function name="PERIODIC_HOOK" language="en_US">
<synopsis>
Execute a periodic dialplan hook into the audio of a call.
</synopsis>
<syntax>
<parameter name="context" required="true">
<para>(On Read Only) Context for the hook extension.</para>
</parameter>
<parameter name="extension" required="true">
<para>(On Read Only) The hook extension.</para>
</parameter>
<parameter name="interval" required="true">
<para>(On Read Only) Number of seconds in between hook runs.
Whole seconds only.</para>
</parameter>
<parameter name="hook_id" required="true">
<para>(On Write Only) The hook ID.</para>
</parameter>
</syntax>
<description>
<para>For example, you could use this function to enable playing
a periodic <literal>beep</literal> sound in a call.</para>
<para/>
<para>To turn on:</para>
<para> Set(BEEPID=${PERIODIC_HOOK(hooks,beep,180)})</para>
<para/>
<para>To turn off:</para>
<para> Set(PERIODIC_HOOK(${BEEPID})=off)</para>
<para/>
<para>To turn back on again later:</para>
<para>Set(PERIODIC_HOOK(${BEEPID})=on)</para>
<para/>
<para>It is important to note that the hook does not actually
run on the channel itself. It runs asynchronously on a new channel.
Any audio generated by the hook gets injected into the call for
the channel PERIODIC_HOOK() was set on.</para>
<para/>
<para>The hook dialplan will have two variables available.
<variable>HOOK_CHANNEL</variable> is the channel the hook is
enabled on. <variable>HOOK_ID</variable> is the hook ID for
enabling or disabling the hook.</para>
</description>
</function>
<function name="PJSIP_ENDPOINT" language="en_US">
<synopsis>
Get information about a PJSIP endpoint
</synopsis>
<syntax>
<parameter name="name" required="true">
<para>The name of the endpoint to query.</para>
</parameter>
<parameter name="field" required="true">
<para>The configuration option for the endpoint to query for.
Supported options are those fields on the
<replaceable>endpoint</replaceable> object in
<filename>pjsip.conf</filename>.</para>
<enumlist>
<configOptionToEnum>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption)"/>
</configOptionToEnum>
</enumlist>
</parameter>
</syntax>
</function>
<function name="SORT" language="en_US">
<synopsis>
Sorts a list of key/vals into a list of keys, based upon the vals.
</synopsis>
<syntax>
<parameter name="keyval" required="true" argsep=":">
<argument name="key1" required="true" />
<argument name="val1" required="true" />
</parameter>
<parameter name="keyvaln" multiple="true" argsep=":">
<argument name="key2" required="true" />
<argument name="val2" required="true" />
</parameter>
</syntax>
<description>
<para>Takes a comma-separated list of keys and values, each separated by a colon, and returns a
comma-separated list of the keys, sorted by their values. Values will be evaluated as
floating-point numbers.</para>
</description>
</function>
<function name="CUT" language="en_US">
<synopsis>
Slices and dices strings, based upon a named delimiter.
</synopsis>
<syntax>
<parameter name="varname" required="true">
<para>Variable you want cut</para>
</parameter>
<parameter name="char-delim" required="true">
<para>Delimiter, defaults to <literal>-</literal></para>
</parameter>
<parameter name="range-spec" required="true">
<para>Number of the field you want (1-based offset), may also be specified as a range (with <literal>-</literal>)
or group of ranges and fields (with <literal>&amp;</literal>)</para>
</parameter>
</syntax>
<description>
<para>Cut out information from a string (<replaceable>varname</replaceable>), based upon a named delimiter.</para>
</description>
</function>
<function name="ENV" language="en_US">
<synopsis>
Gets or sets the environment variable specified.
</synopsis>
<syntax>
<parameter name="varname" required="true">
<para>Environment variable name</para>
</parameter>
</syntax>
<description>
<para>Variables starting with <literal>AST_</literal> are reserved to the system and may not be set.</para>
</description>
</function>
<function name="STAT" language="en_US">
<synopsis>
Does a check on the specified file.
</synopsis>
<syntax>
<parameter name="flag" required="true">
<para>Flag may be one of the following:</para>
<para>d - Checks if the file is a directory.</para>
<para>e - Checks if the file exists.</para>
<para>f - Checks if the file is a regular file.</para>
<para>m - Returns the file mode (in octal)</para>
<para>s - Returns the size (in bytes) of the file</para>
<para>A - Returns the epoch at which the file was last accessed.</para>
<para>C - Returns the epoch at which the inode was last changed.</para>
<para>M - Returns the epoch at which the file was last modified.</para>
</parameter>
<parameter name="filename" required="true" />
</syntax>
<description>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
</function>
<function name="FILE" language="en_US">
<synopsis>
Read or write text file.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="offset">
<para>Maybe specified as any number. If negative, <replaceable>offset</replaceable> specifies the number
of bytes back from the end of the file.</para>
</parameter>
<parameter name="length">
<para>If specified, will limit the length of the data read to that size. If negative,
trims <replaceable>length</replaceable> bytes from the end of the file.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="l">
<para>Line mode: offset and length are assumed to be
measured in lines, instead of byte offsets.</para>
</option>
<option name="a">
<para>In write mode only, the append option is used to
append to the end of the file, instead of overwriting
the existing file.</para>
</option>
<option name="d">
<para>In write mode and line mode only, this option does
not automatically append a newline string to the end of
a value. This is useful for deleting lines, instead of
setting them to blank.</para>
</option>
</optionlist>
</parameter>
<parameter name="format">
<para>The <replaceable>format</replaceable> parameter may be
used to delimit the type of line terminators in line mode.</para>
<optionlist>
<option name="u">
<para>Unix newline format.</para>
</option>
<option name="d">
<para>DOS newline format.</para>
</option>
<option name="m">
<para>Macintosh newline format.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Read and write text file in character and line mode.</para>
<para>Examples:</para>
<para/>
<para>Read mode (byte):</para>
<para> ;reads the entire content of the file.</para>
<para> Set(foo=${FILE(/tmp/test.txt)})</para>
<para> ;reads from the 11th byte to the end of the file (i.e. skips the first 10).</para>
<para> Set(foo=${FILE(/tmp/test.txt,10)})</para>
<para> ;reads from the 11th to 20th byte in the file (i.e. skip the first 10, then read 10 bytes).</para>
<para> Set(foo=${FILE(/tmp/test.txt,10,10)})</para>
<para/>
<para>Read mode (line):</para>
<para> ; reads the 3rd line of the file.</para>
<para> Set(foo=${FILE(/tmp/test.txt,3,1,l)})</para>
<para> ; reads the 3rd and 4th lines of the file.</para>
<para> Set(foo=${FILE(/tmp/test.txt,3,2,l)})</para>
<para> ; reads from the third line to the end of the file.</para>
<para> Set(foo=${FILE(/tmp/test.txt,3,,l)})</para>
<para> ; reads the last three lines of the file.</para>
<para> Set(foo=${FILE(/tmp/test.txt,-3,,l)})</para>
<para> ; reads the 3rd line of a DOS-formatted file.</para>
<para> Set(foo=${FILE(/tmp/test.txt,3,1,l,d)})</para>
<para/>
<para>Write mode (byte):</para>
<para> ; truncate the file and write "bar"</para>
<para> Set(FILE(/tmp/test.txt)=bar)</para>
<para> ; Append "bar"</para>
<para> Set(FILE(/tmp/test.txt,,,a)=bar)</para>
<para> ; Replace the first byte with "bar" (replaces 1 character with 3)</para>
<para> Set(FILE(/tmp/test.txt,0,1)=bar)</para>
<para> ; Replace 10 bytes beginning at the 21st byte of the file with "bar"</para>
<para> Set(FILE(/tmp/test.txt,20,10)=bar)</para>
<para> ; Replace all bytes from the 21st with "bar"</para>
<para> Set(FILE(/tmp/test.txt,20)=bar)</para>
<para> ; Insert "bar" after the 4th character</para>
<para> Set(FILE(/tmp/test.txt,4,0)=bar)</para>
<para/>
<para>Write mode (line):</para>
<para> ; Replace the first line of the file with "bar"</para>
<para> Set(FILE(/tmp/foo.txt,0,1,l)=bar)</para>
<para> ; Replace the last line of the file with "bar"</para>
<para> Set(FILE(/tmp/foo.txt,-1,,l)=bar)</para>
<para> ; Append "bar" to the file with a newline</para>
<para> Set(FILE(/tmp/foo.txt,,,al)=bar)</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
<see-also>
<ref type="function">FILE_COUNT_LINE</ref>
<ref type="function">FILE_FORMAT</ref>
</see-also>
</function>
<function name="FILE_COUNT_LINE" language="en_US">
<synopsis>
Obtains the number of lines of a text file.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="format">
<para>Format may be one of the following:</para>
<optionlist>
<option name="u">
<para>Unix newline format.</para>
</option>
<option name="d">
<para>DOS newline format.</para>
</option>
<option name="m">
<para>Macintosh newline format.</para>
</option>
</optionlist>
<note>
<para>If not specified, an attempt will be made to determine the newline format type.</para>
</note>
</parameter>
</syntax>
<description>
<para>Returns the number of lines, or <literal>-1</literal> on error.</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
<see-also>
<ref type="function">FILE</ref>
<ref type="function">FILE_FORMAT</ref>
</see-also>
</function>
<function name="FILE_FORMAT" language="en_US">
<synopsis>
Return the newline format of a text file.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
</syntax>
<description>
<para>Return the line terminator type:</para>
<para>'u' - Unix "\n" format</para>
<para>'d' - DOS "\r\n" format</para>
<para>'m' - Macintosh "\r" format</para>
<para>'x' - Cannot be determined</para>
<note>
<para>If <literal>live_dangerously</literal> in <literal>asterisk.conf</literal>
is set to <literal>no</literal>, this function can only be executed from the
dialplan, and not directly from external protocols.</para>
</note>
</description>
<see-also>
<ref type="function">FILE</ref>
<ref type="function">FILE_COUNT_LINE</ref>
</see-also>
</function>
<function name="IFMODULE" language="en_US">
<synopsis>
Checks if an Asterisk module is loaded in memory.
</synopsis>
<syntax>
<parameter name="modulename.so" required="true">
<para>Module name complete with <literal>.so</literal></para>
</parameter>
</syntax>
<description>
<para>Checks if a module is loaded. Use the full module name
as shown by the list in <literal>module list</literal>.
Returns <literal>1</literal> if module exists in memory, otherwise <literal>0</literal>
</para>
</description>
</function>
<function name="PITCH_SHIFT" language="en_US">
<synopsis>
Pitch shift both tx and rx audio streams on a channel.
</synopsis>
<syntax>
<parameter name="channel direction" required="true">
<para>Direction can be either <literal>rx</literal>, <literal>tx</literal>, or
<literal>both</literal>. The direction can either be set to a valid floating
point number between 0.1 and 4.0 or one of the enum values listed below. A value
of 1.0 has no effect. Greater than 1 raises the pitch. Lower than 1 lowers
the pitch.</para>
<para>The pitch amount can also be set by the following values</para>
<enumlist>
<enum name = "highest" />
<enum name = "higher" />
<enum name = "high" />
<enum name = "low" />
<enum name = "lower" />
<enum name = "lowest" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Examples:</para>
<para>exten => 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave </para>
<para>exten => 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more </para>
<para>exten => 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch </para>
<para>exten => 1,1,Set(PITCH_SHIFT(rx)=low) ; lowers pitch </para>
<para>exten => 1,1,Set(PITCH_SHIFT(tx)=lower) ; lowers pitch more </para>
<para>exten => 1,1,Set(PITCH_SHIFT(both)=lowest) ; lowers pitch an octave </para>
<para>exten => 1,1,Set(PITCH_SHIFT(rx)=0.8) ; lowers pitch </para>
<para>exten => 1,1,Set(PITCH_SHIFT(tx)=1.5) ; raises pitch </para>
</description>
</function>
<configInfo name="core" language="en_US">
<synopsis>Bucket file API</synopsis>
<configFile name="bucket">
<configObject name="bucket">
<configOption name="scheme">
<synopsis>Scheme in use for bucket</synopsis>
</configOption>
<configOption name="created">
<synopsis>Time at which the bucket was created</synopsis>
</configOption>
<configOption name="modified">
<synopsis>Time at which the bucket was last modified</synopsis>
</configOption>
</configObject>
<configObject name="file">
<configOption name="scheme">
<synopsis>Scheme in use for file</synopsis>
</configOption>
<configOption name="created">
<synopsis>Time at which the file was created</synopsis>
</configOption>
<configOption name="modified">
<synopsis>Time at which the file was last modified</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<managerEvent language="en_US" name="Registry">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when an outbound registration completes.</synopsis>
<syntax>
<parameter name="ChannelType">
<para>The type of channel that was registered (or not).</para>
</parameter>
<parameter name="Username">
<para>The username portion of the registration.</para>
</parameter>
<parameter name="Domain">
<para>The address portion of the registration.</para>
</parameter>
<parameter name="Status">
<para>The status of the registration request.</para>
<enumlist>
<enum name="Registered"/>
<enum name="Unregistered"/>
<enum name="Rejected"/>
<enum name="Failed"/>
</enumlist>
</parameter>
<parameter name="Cause">
<para>What caused the rejection of the request, if available.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="BlindTransfer">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a blind transfer is complete.</synopsis>
<syntax>
<parameter name="Result">
<para>Indicates if the transfer was successful or if it failed.</para>
<enumlist>
<enum name="Fail">
<para>An internal error occurred.</para>
</enum>
<enum name="Invalid">
<para>Invalid configuration for transfer (e.g. Not bridged)</para>
</enum>
<enum name="Not Permitted">
<para>Bridge does not permit transfers</para>
</enum>
<enum name="Success">
<para>Transfer completed successfully</para>
</enum>
</enumlist>
<note>
<para>A result of <literal>Success</literal> does not necessarily mean that a target was succesfully
contacted. It means that a party was succesfully placed into the dialplan at the expected location.</para>
</note>
</parameter>
<channel_snapshot prefix="Transferer"/>
<channel_snapshot prefix="Transferee"/>
<bridge_snapshot/>
<parameter name="IsExternal">
<para>Indicates if the transfer was performed outside of Asterisk. For instance,
a channel protocol native transfer is external. A DTMF transfer is internal.</para>
<enumlist>
<enum name="Yes" />
<enum name="No" />
</enumlist>
</parameter>
<parameter name="Context">
<para>Destination context for the blind transfer.</para>
</parameter>
<parameter name="Extension">
<para>Destination extension for the blind transfer.</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">BlindTransfer</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AttendedTransfer">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when an attended transfer is complete.</synopsis>
<syntax>
<xi:include xpointer="xpointer(docs/managerEvent[@name='BlindTransfer']/managerEventInstance/syntax/parameter[@name='Result'])" />
<channel_snapshot prefix="OrigTransferer"/>
<bridge_snapshot prefix="Orig"/>
<channel_snapshot prefix="SecondTransferer"/>
<bridge_snapshot prefix="Second"/>
<parameter name="DestType">
<para>Indicates the method by which the attended transfer completed.</para>
<enumlist>
<enum name="Bridge">
<para>The transfer was accomplished by merging two bridges into one.</para>
</enum>
<enum name="App">
<para>The transfer was accomplished by having a channel or bridge run a dialplan application.</para>
</enum>
<enum name="Link">
<para>The transfer was accomplished by linking two bridges together using a local channel pair.</para>
</enum>
<enum name="Threeway">
<para>The transfer was accomplished by placing all parties into a threeway call.</para>
</enum>
<enum name="Fail">
<para>The transfer failed.</para>
</enum>
</enumlist>
</parameter>
<parameter name="DestBridgeUniqueid">
<para>Indicates the surviving bridge when bridges were merged to complete the transfer</para>
<note>
<para>This header is only present when <replaceable>DestType</replaceable> is <literal>Bridge</literal> or <literal>Threeway</literal></para>
</note>
</parameter>
<parameter name="DestApp">
<para>Indicates the application that is running when the transfer completes</para>
<note>
<para>This header is only present when <replaceable>DestType</replaceable> is <literal>App</literal></para>
</note>
</parameter>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
<parameter name="DestTransfererChannel">
<para>The name of the surviving transferer channel when a transfer results in a threeway call</para>
<note>
<para>This header is only present when <replaceable>DestType</replaceable> is <literal>Threeway</literal></para>
</note>
</parameter>
<channel_snapshot prefix="Transferee" />
</syntax>
<description>
<para>The headers in this event attempt to describe all the major details of the attended transfer. The two transferer channels
and the two bridges are determined based on their chronological establishment. So consider that Alice calls Bob, and then Alice
transfers the call to Voicemail. The transferer and bridge headers would be arranged as follows:</para>
<para>
<replaceable>OrigTransfererChannel</replaceable>: Alice's channel in the bridge with Bob.</para>
<para>
<replaceable>OrigBridgeUniqueid</replaceable>: The bridge between Alice and Bob.</para>
<para>
<replaceable>SecondTransfererChannel</replaceable>: Alice's channel that called Voicemail.</para>
<para>
<replaceable>SecondBridgeUniqueid</replaceable>: Not present, since a call to Voicemail has no bridge.</para>
<para>Now consider if the order were reversed; instead of having Alice call Bob and transfer him to Voicemail, Alice instead
calls her Voicemail and transfers that to Bob. The transferer and bridge headers would be arranged as follows:</para>
<para>
<replaceable>OrigTransfererChannel</replaceable>: Alice's channel that called Voicemail.</para>
<para>
<replaceable>OrigBridgeUniqueid</replaceable>: Not present, since a call to Voicemail has no bridge.</para>
<para>
<replaceable>SecondTransfererChannel</replaceable>: Alice's channel in the bridge with Bob.</para>
<para>
<replaceable>SecondBridgeUniqueid</replaceable>: The bridge between Alice and Bob.</para>
</description>
<see-also>
<ref type="manager">AtxFer</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="DataGet" language="en_US">
<synopsis>
Retrieve the data api tree.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Path" required="true" />
<parameter name="Search" />
<parameter name="Filter" />
</syntax>
<description>
<para>Retrieve the data api tree.</para>
</description>
</manager>
<manager name="DBGet" language="en_US">
<synopsis>
Get DB Entry.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Family" required="true" />
<parameter name="Key" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="DBPut" language="en_US">
<synopsis>
Put DB entry.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Family" required="true" />
<parameter name="Key" required="true" />
<parameter name="Val" />
</syntax>
<description>
</description>
</manager>
<manager name="DBDel" language="en_US">
<synopsis>
Delete DB entry.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Family" required="true" />
<parameter name="Key" required="true" />
</syntax>
<description>
</description>
</manager>
<manager name="DBDelTree" language="en_US">
<synopsis>
Delete DB Tree.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Family" required="true" />
<parameter name="Key" />
</syntax>
<description>
</description>
</manager>
<configInfo name="cdr" language="en_US">
<synopsis>Call Detail Record configuration</synopsis>
<description>
<para>CDR is Call Detail Record, which provides logging services via a variety of
pluggable backend modules. Detailed call information can be recorded to
databases, files, etc. Useful for billing, fraud prevention, compliance with
Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.</para>
</description>
<configFile name="cdr.conf">
<configObject name="general">
<synopsis>Global settings applied to the CDR engine.</synopsis>
<configOption name="debug">
<synopsis>Enable/disable verbose CDR debugging.</synopsis>
<description>
<para>When set to <literal>True</literal>, verbose updates
of changes in CDR information will be logged. Note that this is only
of use when debugging CDR behavior.</para>
</description>
</configOption>
<configOption name="enable">
<synopsis>Enable/disable CDR logging.</synopsis>
<description>
<para>Define whether or not to use CDR logging. Setting this to "no" will override
any loading of backend CDR modules. Default is "yes".</para>
</description>
</configOption>
<configOption name="unanswered">
<synopsis>Log calls that are never answered and don't set an outgoing party.</synopsis>
<description>
<para>
Define whether or not to log unanswered calls that don't involve an outgoing party. Setting
this to "yes" will make calls to extensions that don't answer and don't set a side B channel
(such as by using the Dial application) receive CDR log entries. If this option is set to
"no", then those log entries will not be created. Unanswered calls which get offered to an
outgoing line will always receive log entries regardless of this option, and that is the
intended behavior.
</para>
</description>
</configOption>
<configOption name="congestion">
<synopsis>Log congested calls.</synopsis>
<description>
<para>Define whether or not to log congested calls. Setting this to "yes" will
report each call that fails to complete due to congestion conditions.</para>
</description>
</configOption>
<configOption name="endbeforehexten">
<synopsis>Don't produce CDRs while executing hangup logic</synopsis>
<description>
<para>As each CDR for a channel is finished, its end time is updated
and the CDR is finalized. When a channel is hung up and hangup
logic is present (in the form of a hangup handler or the
<literal>h</literal> extension), a new CDR is generated for the
channel. Any statistics are gathered from this new CDR. By enabling
this option, no new CDR is created for the dialplan logic that is
executed in <literal>h</literal> extensions or attached hangup handler
subroutines. The default value is <literal>yes</literal>, indicating
that a CDR will be generated during hangup logic.</para>
</description>
</configOption>
<configOption name="initiatedseconds">
<synopsis>Count microseconds for billsec purposes</synopsis>
<description>
<para>Normally, the <literal>billsec</literal> field logged to the CDR backends
is simply the end time (hangup time) minus the answer time in seconds. Internally,
asterisk stores the time in terms of microseconds and seconds. By setting
initiatedseconds to <literal>yes</literal>, you can force asterisk to report any seconds
that were initiated (a sort of round up method). Technically, this is
when the microsecond part of the end time is greater than the microsecond
part of the answer time, then the billsec time is incremented one second.</para>
</description>
</configOption>
<configOption name="batch">
<synopsis>Submit CDRs to the backends for processing in batches</synopsis>
<description>
<para>Define the CDR batch mode, where instead of posting the CDR at the end of
every call, the data will be stored in a buffer to help alleviate load on the
asterisk server.</para>
<warning>
<para>Use of batch mode may result in data loss after unsafe asterisk termination,
i.e., software crash, power failure, kill -9, etc.</para>
</warning>
</description>
</configOption>
<configOption name="size">
<synopsis>The maximum number of CDRs to accumulate before triggering a batch</synopsis>
<description>
<para>Define the maximum number of CDRs to accumulate in the buffer before posting
them to the backend engines. batch must be set to <literal>yes</literal>.</para>
</description>
</configOption>
<configOption name="time">
<synopsis>The maximum time to accumulate CDRs before triggering a batch</synopsis>
<description>
<para>Define the maximum time to accumulate CDRs before posting them in a batch to the
backend engines. If this time limit is reached, then it will post the records, regardless of the value
defined for size. batch must be set to <literal>yes</literal>.</para>
<note>
<para>Time is expressed in seconds.</para>
</note>
</description>
</configOption>
<configOption name="scheduleronly">
<synopsis>Post batched CDRs on their own thread instead of the scheduler</synopsis>
<description>
<para>The CDR engine uses the internal asterisk scheduler to determine when to post
records. Posting can either occur inside the scheduler thread, or a new
thread can be spawned for the submission of every batch. For small batches,
it might be acceptable to just use the scheduler thread, so set this to <literal>yes</literal>.
For large batches, say anything over size=10, a new thread is recommended, so
set this to <literal>no</literal>.</para>
</description>
</configOption>
<configOption name="safeshutdown">
<synopsis>Block shutdown of Asterisk until CDRs are submitted</synopsis>
<description>
<para>When shutting down asterisk, you can block until the CDRs are submitted. If
you don't, then data will likely be lost. You can always check the size of
the CDR batch buffer with the CLI <astcli>cdr status</astcli> command. To enable blocking on
submission of CDR data during asterisk shutdown, set this to <literal>yes</literal>.</para>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="BridgeTechnologyList" language="en_US">
<synopsis>
List available bridging technologies and their statuses.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Returns detailed information about the available bridging technologies.</para>
</description>
<see-also>
<ref type="manager">BridgeTechnologySuspend</ref>
<ref type="manager">BridgeTechnologyUnsuspend</ref>
</see-also>
</manager>
<manager name="BridgeTechnologySuspend" language="en_US">
<synopsis>
Suspend a bridging technology.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeTechnology" required="true">
<para>The name of the bridging technology to suspend.</para>
</parameter>
</syntax>
<description>
<para>Marks a bridging technology as suspended, which prevents subsequently created bridges from using it.</para>
</description>
<see-also>
<ref type="manager">BridgeTechnologySuspend</ref>
<ref type="manager">BridgeTechnologyUnsuspend</ref>
</see-also>
</manager>
<manager name="BridgeTechnologyUnsuspend" language="en_US">
<synopsis>
Unsuspend a bridging technology.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeTechnology" required="true">
<para>The name of the bridging technology to unsuspend.</para>
</parameter>
</syntax>
<description>
<para>Clears a previously suspended bridging technology, which allows subsequently created bridges to use it.</para>
</description>
<see-also>
<ref type="manager">BridgeTechnologyList</ref>
<ref type="manager">BridgeTechnologySuspend</ref>
</see-also>
</manager>
<application name="Bridge" language="en_US">
<synopsis>
Bridge two channels.
</synopsis>
<syntax>
<parameter name="channel" required="true">
<para>The current channel is bridged to the specified <replaceable>channel</replaceable>.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="p">
<para>Play a courtesy tone to <replaceable>channel</replaceable>.</para>
</option>
<option name="F" argsep="^">
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" />
<para>When the bridger hangs up, transfer the <emphasis>bridged</emphasis> party
to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
<note>
<para>This option will override the 'x' option</para>
</note>
</option>
<option name="F">
<para>When the bridger hangs up, transfer the <emphasis>bridged</emphasis> party
to the next priority of the current extension and <emphasis>start</emphasis> execution
at that location.</para>
<note>
<para>Any channel variables you want the called channel to inherit from the caller channel must be
prefixed with one or two underbars ('_').</para>
</note>
<note>
<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
</note>
<note>
<para>This option will override the 'x' option</para>
</note>
</option>
<option name="h">
<para>Allow the called party to hang up by sending the
<replaceable>*</replaceable> DTMF digit.</para>
</option>
<option name="H">
<para>Allow the calling party to hang up by pressing the
<replaceable>*</replaceable> DTMF digit.</para>
</option>
<option name="k">
<para>Allow the called party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="K">
<para>Allow the calling party to enable parking of the call by sending
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
</option>
<option name="L(x[:y][:z])">
<para>Limit the call to <replaceable>x</replaceable> ms. Play a warning
when <replaceable>y</replaceable> ms are left. Repeat the warning every
<replaceable>z</replaceable> ms. The following special variables can be
used with this option:</para>
<variablelist>
<variable name="LIMIT_PLAYAUDIO_CALLER">
<para>Play sounds to the caller. yes|no (default yes)</para>
</variable>
<variable name="LIMIT_PLAYAUDIO_CALLEE">
<para>Play sounds to the callee. yes|no</para>
</variable>
<variable name="LIMIT_TIMEOUT_FILE">
<para>File to play when time is up.</para>
</variable>
<variable name="LIMIT_CONNECT_FILE">
<para>File to play when call begins.</para>
</variable>
<variable name="LIMIT_WARNING_FILE">
<para>File to play as warning if <replaceable>y</replaceable> is
defined. The default is to say the time remaining.</para>
</variable>
</variablelist>
</option>
<option name="S(x)">
<para>Hang up the call after <replaceable>x</replaceable> seconds *after* the called party has answered the call.</para>
</option>
<option name="t">
<para>Allow the called party to transfer the calling party by sending the
DTMF sequence defined in <filename>features.conf</filename>.</para>
</option>
<option name="T">
<para>Allow the calling party to transfer the called party by sending the
DTMF sequence defined in <filename>features.conf</filename>.</para>
</option>
<option name="w">
<para>Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="W">
<para>Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
</option>
<option name="x">
<para>Cause the called party to be hung up after the bridge, instead of being
restarted in the dialplan.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Allows the ability to bridge two channels via the dialplan.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="BRIDGERESULT">
<para>The result of the bridge attempt as a text string.</para>
<value name="SUCCESS" />
<value name="FAILURE" />
<value name="LOOP" />
<value name="NONEXISTENT" />
<value name="INCOMPATIBLE" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="manager">Bridge</ref>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeEnter</ref>
</see-also>
</application>
<manager name="Bridge" language="en_US">
<synopsis>
Bridge two channels already in the PBX.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel1" required="true">
<para>Channel to Bridge to Channel2.</para>
</parameter>
<parameter name="Channel2" required="true">
<para>Channel to Bridge to Channel1.</para>
</parameter>
<parameter name="Tone">
<para>Play courtesy tone to Channel 2.</para>
<enumlist>
<enum name="no" />
<enum name="Channel1" />
<enum name="Channel2" />
<enum name="Both" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Bridge together two channels already in the PBX.</para>
</description>
<see-also>
<ref type="application">Bridge</ref>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeEnter</ref>
<ref type="manager">BridgeDestroy</ref>
<ref type="manager">BridgeInfo</ref>
<ref type="manager">BridgeKick</ref>
<ref type="manager">BridgeList</ref>
</see-also>
</manager>
<configInfo name="features" language="en_US">
<synopsis>Features Configuration</synopsis>
<configFile name="features.conf">
<configObject name="globals">
<synopsis>
</synopsis>
<configOption name="featuredigittimeout" default="1000">
<synopsis>Milliseconds allowed between digit presses when entering a feature code.</synopsis>
</configOption>
<configOption name="courtesytone">
<synopsis>Sound to play when automon or automixmon is activated</synopsis>
</configOption>
<configOption name="recordingfailsound">
<synopsis>Sound to play when automon or automixmon is attempted but fails to start</synopsis>
</configOption>
<configOption name="transferdigittimeout" default="3">
<synopsis>Seconds allowed between digit presses when dialing a transfer destination</synopsis>
</configOption>
<configOption name="atxfernoanswertimeout" default="15">
<synopsis>Seconds to wait for attended transfer destination to answer</synopsis>
</configOption>
<configOption name="atxferdropcall" default="no">
<synopsis>Hang up the call entirely if the attended transfer fails</synopsis>
<description>
<para>When this option is set to <literal>no</literal>, then Asterisk will attempt to
re-call the transferrer if the call to the transfer target fails. If the call to the
transferrer fails, then Asterisk will wait <replaceable>atxferloopdelay</replaceable>
milliseconds and then attempt to dial the transfer target again. This process will
repeat until <replaceable>atxfercallbackretries</replaceable> attempts to re-call
the transferrer have occurred.</para>
<para>When this option is set to <literal>yes</literal>, then Asterisk will not attempt
to re-call the transferrer if the call to the transfer target fails. Asterisk will instead
hang up all channels involved in the transfer.</para>
</description>
</configOption>
<configOption name="atxferloopdelay" default="10">
<synopsis>Seconds to wait between attempts to re-dial transfer destination</synopsis>
<see-also>
<ref type="configOption">atxferdropcall</ref>
</see-also>
</configOption>
<configOption name="atxfercallbackretries" default="2">
<synopsis>Number of times to re-attempt dialing a transfer destination</synopsis>
<see-also>
<ref type="configOption">atxferdropcall</ref>
</see-also>
</configOption>
<configOption name="xfersound" default="beep">
<synopsis>Sound to play to during transfer and transfer-like operations.</synopsis>
<description>
<para>This sound will play to the transferrer and transfer target channels when
an attended transfer completes. This sound is also played to channels when performing
an AMI <literal>Bridge</literal> action.</para>
</description>
</configOption>
<configOption name="xferfailsound" default="beeperr">
<synopsis>Sound to play to a transferee when a transfer fails</synopsis>
</configOption>
<configOption name="atxferabort" default="*1">
<synopsis>Digits to dial to abort an attended transfer attempt</synopsis>
<description>
<para>This option is only available to the transferrer during an attended
transfer operation. Aborting a transfer results in the transfer being cancelled and
the original parties in the call being re-bridged.</para>
</description>
</configOption>
<configOption name="atxfercomplete" default="*2">
<synopsis>Digits to dial to complete an attended transfer</synopsis>
<description>
<para>This option is only available to the transferrer during an attended
transfer operation. Completing the transfer with a DTMF sequence is functionally
equivalent to hanging up the transferrer channel during an attended transfer. The
result is that the transfer target and transferees are bridged.</para>
</description>
</configOption>
<configOption name="atxferthreeway" default="*3">
<synopsis>Digits to dial to change an attended transfer into a three-way call</synopsis>
<description>
<para>This option is only available to the transferrer during an attended
transfer operation. Pressing this DTMF sequence will result in the transferrer,
the transferees, and the transfer target all being in a single bridge together.</para>
</description>
</configOption>
<configOption name="atxferswap" default="*4">
<synopsis>Digits to dial to toggle who the transferrer is currently bridged to during an attended transfer</synopsis>
<description>
<para>This option is only available to the transferrer during an attended
transfer operation. Pressing this DTMF sequence will result in the transferrer swapping
which party he is bridged with. For instance, if the transferrer is currently bridged with
the transfer target, then pressing this DTMF sequence will cause the transferrer to be
bridged with the transferees.</para>
</description>
</configOption>
<configOption name="pickupexten" default="*8">
<synopsis>Digits used for picking up ringing calls</synopsis>
<description>
<para>In order for the pickup attempt to be successful, the party attempting to
pick up the call must either have a <replaceable>namedpickupgroup</replaceable> in
common with a ringing party's <replaceable>namedcallgroup</replaceable> or must
have a <replaceable>pickupgroup</replaceable> in common with a ringing party's
<replaceable>callgroup</replaceable>.</para>
</description>
</configOption>
<configOption name="pickupsound">
<synopsis>Sound to play to picker when a call is picked up</synopsis>
</configOption>
<configOption name="pickupfailsound">
<synopsis>Sound to play to picker when a call cannot be picked up</synopsis>
</configOption>
<configOption name="transferdialattempts" default="3">
<synopsis>Number of dial attempts allowed when attempting a transfer</synopsis>
</configOption>
<configOption name="transferretrysound" default="pbx-invalid">
<synopsis>Sound that is played when an incorrect extension is dialed and the transferer should try again.</synopsis>
</configOption>
<configOption name="transferinvalidsound" default="privacy-incorrect">
<synopsis>Sound that is played when an incorrect extension is dialed and the transferer has no attempts remaining.</synopsis>
</configOption>
</configObject>
<configObject name="featuremap">
<synopsis>DTMF options that can be triggered during bridged calls</synopsis>
<configOption name="atxfer">
<synopsis>DTMF sequence to initiate an attended transfer</synopsis>
<description>
<para>The transferee parties will be placed on hold and the
transferrer may dial an extension to reach a transfer target. During an
attended transfer, the transferrer may consult with the transfer target
before completing the transfer. Once the transferrer has hung up or pressed
the <replaceable>atxfercomplete</replaceable> DTMF sequence, then the transferees
and transfer target will be bridged.</para>
</description>
</configOption>
<configOption name="blindxfer" default="#">
<synopsis>DTMF sequence to initiate a blind transfer</synopsis>
<description>
<para>The transferee parties will be placed on hold and the
transferrer may dial an extension to reach a transfer target. During a
blind transfer, as soon as the transfer target is dialed, the transferrer
is hung up.</para>
</description>
</configOption>
<configOption name="disconnect" default="*">
<synopsis>DTMF sequence to disconnect the current call</synopsis>
<description>
<para>Entering this DTMF sequence will cause the bridge to end, no
matter the number of parties present</para>
</description>
</configOption>
<configOption name="parkcall">
<synopsis>DTMF sequence to park a call</synopsis>
<description>
<para>The parking lot used to park the call is determined by using either the
<replaceable>PARKINGLOT</replaceable> channel variable or a configured value on
the channel (provided by the channel driver) if the variable is not present. If
no configured value on the channel is present, then <literal>"default"</literal>
is used. The call is parked in the next available space in the parking lot.</para>
</description>
</configOption>
<configOption name="automon">
<synopsis>DTMF sequence to start or stop monitoring a call</synopsis>
<description>
<para>This will cause the channel that pressed the DTMF sequence
to be monitored by the <literal>Monitor</literal> application. The
format for the recording is determined by the <replaceable>TOUCH_MONITOR_FORMAT</replaceable>
channel variable. If this variable is not specified, then <literal>wav</literal> is the
default. The filename is constructed in the following manner:</para>
<para> prefix-timestamp-filename</para>
<para>where prefix is either the value of the <replaceable>TOUCH_MONITOR_PREFIX</replaceable>
channel variable or <literal>auto</literal> if the variable is not set. The timestamp
is a UNIX timestamp. The filename is either the value of the <replaceable>TOUCH_MONITOR</replaceable>
channel variable or the callerID of the channels if the variable is not set.</para>
</description>
</configOption>
<configOption name="automixmon">
<synopsis>DTMF sequence to start or stop mixmonitoring a call </synopsis>
<description>
<para>Operation of the automixmon is similar to the <literal> automon </literal>
feature, with the following exceptions:
<replaceable>TOUCH_MIXMONITOR</replaceable> is used in place of <replaceable>TOUCH_MONITOR</replaceable>
<replaceable>TOUCH_MIXMONITOR_FORMAT</replaceable> is used in place of <replaceable>TOUCH_MIXMONITOR</replaceable>
There is no equivalent for <replaceable>TOUCH_MONITOR_PREFIX</replaceable>. <literal>"auto"</literal> is always how the filename begins.</para>
</description>
<see-also>
<ref type="configOption">automon</ref>
</see-also>
</configOption>
</configObject>
<configObject name="applicationmap">
<synopsis>Section for defining custom feature invocations during a call</synopsis>
<description>
<para>The applicationmap is an area where new custom features can be created. Items
defined in the applicationmap are not automatically accessible to bridged parties. Access
to the individual items is controled using the <replaceable>DYNAMIC_FEATURES</replaceable> channel variable.
The <replaceable>DYNAMIC_FEATURES</replaceable> is a <literal>#</literal> separated list of
either applicationmap item names or featuregroup names.</para>
</description>
<configOption name="^.*$" regex="true">
<synopsis>A custom feature to invoke during a bridged call</synopsis>
<description>
<para>Each item listed here is a comma-separated list of parameters that determine
how a feature may be invoked during a call</para>
<para> Example:</para>
<para> eggs = *5,self,Playback(hello-world),default</para>
<para>This would create a feature called <literal>eggs</literal> that could be invoked
during a call by pressing the <literal>*5</literal>. The party that presses the DTMF
sequence would then trigger the <literal>Playback</literal> application to play the
<literal>hello-world</literal> file. The application invocation would happen on the
party that pressed the DTMF sequence since <literal>self</literal> is specified. The
other parties in the bridge would hear the <literal>default</literal> music on hold
class during the playback.</para>
<para>In addition to the syntax outlined in this documentation, a backwards-compatible alternative
is also allowed. The following applicationmap lines are functionally identical:</para>
<para> eggs = *5,self,Playback(hello-world),default</para>
<para> eggs = *5,self,Playback,hello-world,default</para>
<para> eggs = *5,self,Playback,"hello-world",default</para>
</description>
<syntax argsep=",">
<parameter name="dtmf" required="true">
<para>The DTMF sequence used to trigger the option</para>
</parameter>
<parameter name="activate_on" required="true">
<para>The party that the feature will be invoked on</para>
<optionlist>
<option name="self">
<para>Feature is invoked on party that presses the DTMF sequence</para>
</option>
<option name="peer">
<para>Feature is invoked on other parties in the bridge</para>
</option>
</optionlist>
</parameter>
<parameter name="app" required="true">
<para>The dialplan application to run when the DTMF sequence is pressed</para>
<argument name="app_args" required="false">
<para>The arguments to the dialplan application to run</para>
</argument>
</parameter>
<parameter name="moh_class" required="false">
<para>Music on hold class to play to bridge participants that are not the target of the application invocation</para>
</parameter>
</syntax>
</configOption>
</configObject>
<configObject name="featuregroup">
<synopsis>Groupings of items from the applicationmap</synopsis>
<description>
<para>Feature groups allow for multiple applicationmap items to be
grouped together. Like with individual applicationmap items, feature groups
can be part of the <replaceable>DYNAMIC_FEATURES</replaceable> channel variable.
In addition to creating groupings, the feature group section allows for the
DTMF sequence used to invoke an applicationmap item to be overridden with
a different sequence.</para>
</description>
<configOption name="^.*$" regex="true">
<synopsis>Applicationmap item to place in the feature group</synopsis>
<description>
<para>Each item here must be a name of an item in the applicationmap. The
argument may either be a new DTMF sequence to use for the item or it
may be left blank in order to use the DTMF sequence specified in the
applicationmap. For example:</para>
<para> eggs => *1</para>
<para> bacon =></para>
<para>would result in the applicationmap items <literal>eggs</literal> and
<literal>bacon</literal> being in the featuregroup. The former would have its
default DTMF trigger overridden with <literal>*1</literal> and the latter would
have the DTMF value specified in the applicationmap.</para>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<function name="FEATURE" language="en_US">
<synopsis>
Get or set a feature option on a channel.
</synopsis>
<syntax>
<parameter name="option_name" required="true">
<para>The allowed values are:</para>
<enumlist>
<enum name="inherit">
<para>Inherit feature settings made in FEATURE or FEATUREMAP to child channels.</para>
</enum>
<enum name="featuredigittimeout">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='featuredigittimeout']/synopsis/text())" />
</para>
</enum>
<enum name="transferdigittimeout">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='transferdigittimeout']/synopsis/text())" />
</para>
</enum>
<enum name="atxfernoanswertimeout">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxfernoanswertimeout']/synopsis/text())" />
</para>
</enum>
<enum name="atxferdropcall">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxferdropcall']/synopsis/text())" />
</para>
</enum>
<enum name="atxferloopdelay">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxferloopdelay']/synopsis/text())" />
</para>
</enum>
<enum name="atxfercallbackretries">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxfercallbackretries']/synopsis/text())" />
</para>
</enum>
<enum name="xfersound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='xfersound']/synopsis/text())" />
</para>
</enum>
<enum name="xferfailsound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='xferfailsound']/synopsis/text())" />
</para>
</enum>
<enum name="atxferabort">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxferabort']/synopsis/text())" />
</para>
</enum>
<enum name="atxfercomplete">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxfercomplete']/synopsis/text())" />
</para>
</enum>
<enum name="atxferthreeway">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='atxferthreeway']/synopsis/text())" />
</para>
</enum>
<enum name="pickupexten">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='pickupexten']/synopsis/text())" />
</para>
</enum>
<enum name="pickupsound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='pickupsound']/synopsis/text())" />
</para>
</enum>
<enum name="pickupfailsound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='pickupfailsound']/synopsis/text())" />
</para>
</enum>
<enum name="courtesytone">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='courtesytone']/synopsis/text())" />
</para>
</enum>
<enum name="recordingfailsound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='recordingfailsound']/synopsis/text())" />
</para>
</enum>
<enum name="transferdialattempts">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='transferdialattempts']/synopsis/text())" />
</para>
</enum>
<enum name="transferretrysound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='transferretrysound']/synopsis/text())" />
</para>
</enum>
<enum name="transferinvalidsound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='features']/configFile[@name='features.conf']/configObject[@name='globals']/configOption[@name='transferinvalidsound']/synopsis/text())" />
</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>When this function is used as a read, it will get the current
value of the specified feature option for this channel. It will be
the value of this option configured in features.conf if a channel specific
value has not been set. This function can also be used to set a channel
specific value for the supported feature options.</para>
</description>
<see-also>
<ref type="function">FEATUREMAP</ref>
</see-also>
</function>
<function name="FEATUREMAP" language="en_US">
<synopsis>
Get or set a feature map to a given value on a specific channel.
</synopsis>
<syntax>
<parameter name="feature_name" required="true">
<para>The allowed values are:</para>
<enumlist>
<enum name="atxfer">
<para>Attended Transfer</para>
</enum>
<enum name="blindxfer">
<para>Blind Transfer</para>
</enum>
<enum name="automon">
<para>Auto Monitor</para>
</enum>
<enum name="disconnect">
<para>Call Disconnect</para>
</enum>
<enum name="parkcall">
<para>Park Call</para>
</enum>
<enum name="automixmon">
<para>Auto MixMonitor</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>When this function is used as a read, it will get the current
digit sequence mapped to the specified feature for this channel. This
value will be the one configured in features.conf if a channel specific
value has not been set. This function can also be used to set a channel
specific value for a feature mapping.</para>
</description>
<see-also>
<ref type="function">FEATURE</ref>
</see-also>
</function>
<configInfo name="udptl" language="en_US">
<configFile name="udptl.conf">
<configObject name="global">
<synopsis>Global options for configuring UDPTL</synopsis>
<configOption name="udptlstart">
<synopsis>The start of the UDPTL port range</synopsis>
</configOption>
<configOption name="udptlend">
<synopsis>The end of the UDPTL port range</synopsis>
</configOption>
<configOption name="udptlchecksums">
<synopsis>Whether to enable or disable UDP checksums on UDPTL traffic</synopsis>
</configOption>
<configOption name="udptlfecentries">
<synopsis>The number of error correction entries in a UDPTL packet</synopsis>
</configOption>
<configOption name="udptlfecspan">
<synopsis>The span over which parity is calculated for FEC in a UDPTL packet</synopsis>
</configOption>
<configOption name="use_even_ports">
<synopsis>Whether to only use even-numbered UDPTL ports</synopsis>
</configOption>
<configOption name="t38faxudpec">
<synopsis>Removed</synopsis>
</configOption>
<configOption name="t38faxmaxdatagram">
<synopsis>Removed</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<managerEvent language="en_US" name="BridgeCreate">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a bridge is created.</synopsis>
<syntax>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">BridgeDestroy</ref>
<ref type="managerEvent">BridgeEnter</ref>
<ref type="managerEvent">BridgeLeave</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="BridgeDestroy">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a bridge is destroyed.</synopsis>
<syntax>
<bridge_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeEnter</ref>
<ref type="managerEvent">BridgeLeave</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="BridgeEnter">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel enters a bridge.</synopsis>
<syntax>
<bridge_snapshot/>
<channel_snapshot/>
<parameter name="SwapUniqueid">
<para>The uniqueid of the channel being swapped out of the bridge</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeDestroy</ref>
<ref type="managerEvent">BridgeLeave</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="BridgeLeave">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel leaves a bridge.</synopsis>
<syntax>
<bridge_snapshot/>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeDestroy</ref>
<ref type="managerEvent">BridgeEnter</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="BridgeVideoSourceUpdate">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when the channel that is the source of video in a bridge changes.</synopsis>
<syntax>
<bridge_snapshot/>
<parameter name="BridgePreviousVideoSource">
<para>The unique ID of the channel that was the video source.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">BridgeCreate</ref>
<ref type="managerEvent">BridgeDestroy</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="BridgeList" language="en_US">
<synopsis>
Get a list of bridges in the system.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeType">
<para>Optional type for filtering the resulting list of bridges.</para>
</parameter>
</syntax>
<description>
<para>Returns a list of bridges, optionally filtering on a bridge type.</para>
</description>
<see-also>
<ref type="manager">Bridge</ref>
<ref type="manager">BridgeDestroy</ref>
<ref type="manager">BridgeInfo</ref>
<ref type="manager">BridgeKick</ref>
</see-also>
</manager>
<manager name="BridgeInfo" language="en_US">
<synopsis>
Get information about a bridge.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeUniqueid" required="true">
<para>The unique ID of the bridge about which to retrieve information.</para>
</parameter>
</syntax>
<description>
<para>Returns detailed information about a bridge and the channels in it.</para>
</description>
<see-also>
<ref type="manager">Bridge</ref>
<ref type="manager">BridgeDestroy</ref>
<ref type="manager">BridgeKick</ref>
<ref type="manager">BridgeList</ref>
</see-also>
<responses>
<list-elements>
<managerEvent language="en_US" name="BridgeInfoChannel">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Information about a channel in a bridge.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
</managerEventInstance>
</managerEvent>
</list-elements>
<managerEvent language="en_US" name="BridgeInfoComplete">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Information about a bridge.</synopsis>
<syntax>
<bridge_snapshot/>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<manager name="BridgeDestroy" language="en_US">
<synopsis>
Destroy a bridge.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeUniqueid" required="true">
<para>The unique ID of the bridge to destroy.</para>
</parameter>
</syntax>
<description>
<para>Deletes the bridge, causing channels to continue or hang up.</para>
</description>
<see-also>
<ref type="manager">Bridge</ref>
<ref type="manager">BridgeInfo</ref>
<ref type="manager">BridgeKick</ref>
<ref type="manager">BridgeList</ref>
<ref type="managerEvent">BridgeDestroy</ref>
</see-also>
</manager>
<manager name="BridgeKick" language="en_US">
<synopsis>
Kick a channel from a bridge.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="BridgeUniqueid" required="false">
<para>The unique ID of the bridge containing the channel to
destroy. This parameter can be omitted, or supplied to insure
that the channel is not removed from the wrong bridge.</para>
</parameter>
<parameter name="Channel" required="true">
<para>The channel to kick out of a bridge.</para>
</parameter>
</syntax>
<description>
<para>The channel is removed from the bridge.</para>
</description>
<see-also>
<ref type="manager">Bridge</ref>
<ref type="manager">BridgeDestroy</ref>
<ref type="manager">BridgeInfo</ref>
<ref type="manager">BridgeList</ref>
<ref type="managerEvent">BridgeLeave</ref>
</see-also>
</manager>
<function name="MESSAGE" language="en_US">
<synopsis>
Create a message or read fields from a message.
</synopsis>
<syntax argsep="/">
<parameter name="argument" required="true">
<para>Field of the message to get or set.</para>
<enumlist>
<enum name="to">
<para>Read-only. The destination of the message. When processing an
incoming message, this will be set to the destination listed as
the recipient of the message that was received by Asterisk.</para>
</enum>
<enum name="from">
<para>Read-only. The source of the message. When processing an
incoming message, this will be set to the source of the message.</para>
</enum>
<enum name="custom_data">
<para>Write-only. Mark or unmark all message headers for an outgoing
message. The following values can be set:</para>
<enumlist>
<enum name="mark_all_outbound">
<para>Mark all headers for an outgoing message.</para>
</enum>
<enum name="clear_all_outbound">
<para>Unmark all headers for an outgoing message.</para>
</enum>
</enumlist>
</enum>
<enum name="body">
<para>Read/Write. The message body. When processing an incoming
message, this includes the body of the message that Asterisk
received. When MessageSend() is executed, the contents of this
field are used as the body of the outgoing message. The body
will always be UTF-8.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function will read from or write a value to a text message.
It is used both to read the data out of an incoming message, as well as
modify or create a message that will be sent outbound.</para>
</description>
<see-also>
<ref type="application">MessageSend</ref>
</see-also>
</function>
<function name="MESSAGE_DATA" language="en_US">
<synopsis>
Read or write custom data attached to a message.
</synopsis>
<syntax argsep="/">
<parameter name="argument" required="true">
<para>Field of the message to get or set.</para>
</parameter>
</syntax>
<description>
<para>This function will read from or write a value to a text message.
It is used both to read the data out of an incoming message, as well as
modify a message that will be sent outbound.</para>
<note>
<para>If you want to set an outbound message to carry data in the
current message, do
Set(MESSAGE_DATA(<replaceable>key</replaceable>)=${MESSAGE_DATA(<replaceable>key</replaceable>)}).</para>
</note>
</description>
<see-also>
<ref type="application">MessageSend</ref>
</see-also>
</function>
<application name="MessageSend" language="en_US">
<synopsis>
Send a text message.
</synopsis>
<syntax>
<parameter name="to" required="true">
<para>A To URI for the message.</para>
<xi:include xpointer="xpointer(/docs/info[@name='MessageToInfo'])" />
</parameter>
<parameter name="from" required="false">
<para>A From URI for the message if needed for the
message technology being used to send this message. This can be a
SIP(S) URI, such as <literal>Alice &lt;sip:alice@atlanta.com&gt;</literal>,
a string in the format <literal>alice@atlanta.com</literal>, or simply
a username such as <literal>alice</literal>.</para>
</parameter>
</syntax>
<description>
<para>Send a text message. The body of the message that will be
sent is what is currently set to <literal>MESSAGE(body)</literal>.
The technology chosen for sending the message is determined
based on a prefix to the <literal>to</literal> parameter.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="MESSAGE_SEND_STATUS">
<para>This is the message delivery status returned by this application.</para>
<value name="INVALID_PROTOCOL">
No handler for the technology part of the URI was found.
</value>
<value name="INVALID_URI">
The protocol handler reported that the URI was not valid.
</value>
<value name="SUCCESS">
Successfully passed on to the protocol handler, but delivery has not necessarily been guaranteed.
</value>
<value name="FAILURE">
The protocol handler reported that it was unabled to deliver the message for some reason.
</value>
</variable>
</variablelist>
</description>
</application>
<manager name="MessageSend" language="en_US">
<synopsis>
Send an out of call message to an endpoint.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="To" required="true">
<para>The URI the message is to be sent to.</para>
<xi:include xpointer="xpointer(/docs/info[@name='MessageToInfo'])" />
</parameter>
<parameter name="From">
<para>A From URI for the message if needed for the
message technology being used to send this message.</para>
<xi:include xpointer="xpointer(/docs/info[@name='MessageFromInfo'])" />
</parameter>
<parameter name="Body">
<para>The message body text. This must not contain any newlines as that
conflicts with the AMI protocol.</para>
</parameter>
<parameter name="Base64Body">
<para>Text bodies requiring the use of newlines have to be base64 encoded
in this field. Base64Body will be decoded before being sent out.
Base64Body takes precedence over Body.</para>
</parameter>
<parameter name="Variable">
<para>Message variable to set, multiple Variable: headers are
allowed. The header value is a comma separated list of
name=value pairs.</para>
</parameter>
</syntax>
</manager>
<managerEvent language="en_US" name="UserEvent">
<managerEventInstance class="EVENT_FLAG_USER">
<synopsis>A user defined event raised from the dialplan.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="UserEvent">
<para>The event name, as specified in the dialplan.</para>
</parameter>
</syntax>
<description>
<para>Event may contain additional arbitrary parameters in addition to optional bridge and endpoint snapshots. Multiple snapshots of the same type are prefixed with a numeric value.</para>
</description>
<see-also>
<ref type="application">UserEvent</ref>
<ref type="managerEvent">UserEvent</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<configInfo name="stasis" language="en_US">
<configFile name="stasis.conf">
<configObject name="threadpool">
<synopsis>Settings that configure the threadpool Stasis uses to deliver some messages.</synopsis>
<configOption name="initial_size" default="5">
<synopsis>Initial number of threads in the message bus threadpool.</synopsis>
</configOption>
<configOption name="idle_timeout_sec" default="20">
<synopsis>Number of seconds before an idle thread is disposed of.</synopsis>
</configOption>
<configOption name="max_size" default="50">
<synopsis>Maximum number of threads in the threadpool.</synopsis>
</configOption>
</configObject>
<configObject name="declined_message_types">
<synopsis>Stasis message types for which to decline creation.</synopsis>
<configOption name="decline">
<synopsis>The message type to decline.</synopsis>
<description>
<para>This configuration option defines the name of the Stasis
message type that Asterisk is forbidden from creating and can be
specified as many times as necessary to achieve the desired result.</para>
<enumlist>
<enum name="stasis_app_recording_snapshot_type" />
<enum name="stasis_app_playback_snapshot_type" />
<enum name="stasis_test_message_type" />
<enum name="confbridge_start_type" />
<enum name="confbridge_end_type" />
<enum name="confbridge_join_type" />
<enum name="confbridge_leave_type" />
<enum name="confbridge_start_record_type" />
<enum name="confbridge_stop_record_type" />
<enum name="confbridge_mute_type" />
<enum name="confbridge_unmute_type" />
<enum name="confbridge_talking_type" />
<enum name="cel_generic_type" />
<enum name="ast_bridge_snapshot_type" />
<enum name="ast_bridge_merge_message_type" />
<enum name="ast_channel_entered_bridge_type" />
<enum name="ast_channel_left_bridge_type" />
<enum name="ast_blind_transfer_type" />
<enum name="ast_attended_transfer_type" />
<enum name="ast_endpoint_snapshot_type" />
<enum name="ast_endpoint_state_type" />
<enum name="ast_device_state_message_type" />
<enum name="ast_test_suite_message_type" />
<enum name="ast_mwi_state_type" />
<enum name="ast_mwi_vm_app_type" />
<enum name="ast_format_register_type" />
<enum name="ast_format_unregister_type" />
<enum name="ast_manager_get_generic_type" />
<enum name="ast_parked_call_type" />
<enum name="ast_channel_snapshot_type" />
<enum name="ast_channel_dial_type" />
<enum name="ast_channel_varset_type" />
<enum name="ast_channel_hangup_request_type" />
<enum name="ast_channel_dtmf_begin_type" />
<enum name="ast_channel_dtmf_end_type" />
<enum name="ast_channel_hold_type" />
<enum name="ast_channel_unhold_type" />
<enum name="ast_channel_chanspy_start_type" />
<enum name="ast_channel_chanspy_stop_type" />
<enum name="ast_channel_fax_type" />
<enum name="ast_channel_hangup_handler_type" />
<enum name="ast_channel_moh_start_type" />
<enum name="ast_channel_moh_stop_type" />
<enum name="ast_channel_monitor_start_type" />
<enum name="ast_channel_monitor_stop_type" />
<enum name="ast_channel_agent_login_type" />
<enum name="ast_channel_agent_logoff_type" />
<enum name="ast_channel_talking_start" />
<enum name="ast_channel_talking_stop" />
<enum name="ast_security_event_type" />
<enum name="ast_named_acl_change_type" />
<enum name="ast_local_bridge_type" />
<enum name="ast_local_optimization_begin_type" />
<enum name="ast_local_optimization_end_type" />
<enum name="stasis_subscription_change_type" />
<enum name="ast_multi_user_event_type" />
<enum name="stasis_cache_clear_type" />
<enum name="stasis_cache_update_type" />
<enum name="ast_network_change_type" />
<enum name="ast_system_registry_type" />
<enum name="ast_cc_available_type" />
<enum name="ast_cc_offertimerstart_type" />
<enum name="ast_cc_requested_type" />
<enum name="ast_cc_requestacknowledged_type" />
<enum name="ast_cc_callerstopmonitoring_type" />
<enum name="ast_cc_callerstartmonitoring_type" />
<enum name="ast_cc_callerrecalling_type" />
<enum name="ast_cc_recallcomplete_type" />
<enum name="ast_cc_failure_type" />
<enum name="ast_cc_monitorfailed_type" />
<enum name="ast_presence_state_message_type" />
<enum name="ast_rtp_rtcp_sent_type" />
<enum name="ast_rtp_rtcp_received_type" />
<enum name="ast_call_pickup_type" />
<enum name="aoc_s_type" />
<enum name="aoc_d_type" />
<enum name="aoc_e_type" />
<enum name="dahdichannel_type" />
<enum name="mcid_type" />
<enum name="session_timeout_type" />
<enum name="cdr_read_message_type" />
<enum name="cdr_write_message_type" />
<enum name="cdr_prop_write_message_type" />
<enum name="corosync_ping_message_type" />
<enum name="agi_exec_start_type" />
<enum name="agi_exec_end_type" />
<enum name="agi_async_start_type" />
<enum name="agi_async_exec_type" />
<enum name="agi_async_end_type" />
<enum name="queue_caller_join_type" />
<enum name="queue_caller_leave_type" />
<enum name="queue_caller_abandon_type" />
<enum name="queue_member_status_type" />
<enum name="queue_member_added_type" />
<enum name="queue_member_removed_type" />
<enum name="queue_member_pause_type" />
<enum name="queue_member_penalty_type" />
<enum name="queue_member_ringinuse_type" />
<enum name="queue_agent_called_type" />
<enum name="queue_agent_connect_type" />
<enum name="queue_agent_complete_type" />
<enum name="queue_agent_dump_type" />
<enum name="queue_agent_ringnoanswer_type" />
<enum name="meetme_join_type" />
<enum name="meetme_leave_type" />
<enum name="meetme_end_type" />
<enum name="meetme_mute_type" />
<enum name="meetme_talking_type" />
<enum name="meetme_talk_request_type" />
<enum name="appcdr_message_type" />
<enum name="forkcdr_message_type" />
<enum name="cdr_sync_message_type" />
</enumlist>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="LocalOptimizeAway" language="en_US">
<synopsis>
Optimize away a local channel when possible.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The channel name to optimize away.</para>
</parameter>
</syntax>
<description>
<para>A local channel created with "/n" will not automatically optimize away.
Calling this command on the local channel will clear that flag and allow
it to optimize away if it's bridged or when it becomes bridged.</para>
</description>
</manager>
<managerEvent language="en_US" name="LocalBridge">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel form a bridge.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
<parameter name="Context">
<para>The context in the dialplan that Channel2 starts in.</para>
</parameter>
<parameter name="Exten">
<para>The extension in the dialplan that Channel2 starts in.</para>
</parameter>
<parameter name="LocalOptimization">
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="LocalOptimizationBegin">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel begin to optimize
themselves out of the media path.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
<channel_snapshot prefix="Source"/>
<parameter name="DestUniqueId">
<para>The unique ID of the bridge into which the local channel is optimizing.</para>
</parameter>
<parameter name="Id">
<para>Identification for the optimization operation.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">LocalOptimizationEnd</ref>
<ref type="manager">LocalOptimizeAway</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="LocalOptimizationEnd">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel have finished optimizing
themselves out of the media path.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
<parameter name="Success">
<para>Indicates whether the local optimization succeeded.</para>
</parameter>
<parameter name="Id">
<para>Identification for the optimization operation. Matches the <replaceable>Id</replaceable>
from a previous <literal>LocalOptimizationBegin</literal>
</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">LocalOptimizationBegin</ref>
<ref type="manager">LocalOptimizeAway</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="FailedACL">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request violates an ACL check.</synopsis>
<syntax>
<parameter name="EventTV">
<para>The time the event was detected.</para>
</parameter>
<parameter name="Severity">
<para>A relative severity of the security event.</para>
<enumlist>
<enum name="Informational"/>
<enum name="Error"/>
</enumlist>
</parameter>
<parameter name="Service">
<para>The Asterisk service that raised the security event.</para>
</parameter>
<parameter name="EventVersion">
<para>The version of this event.</para>
</parameter>
<parameter name="AccountID">
<para>The Service account associated with the security event
notification.</para>
</parameter>
<parameter name="SessionID">
<para>A unique identifier for the session in the service
that raised the event.</para>
</parameter>
<parameter name="LocalAddress">
<para>The address of the Asterisk service that raised the
security event.</para>
</parameter>
<parameter name="RemoteAddress">
<para>The remote address of the entity that caused the
security event to be raised.</para>
</parameter>
<parameter name="Module" required="false">
<para>If available, the name of the module that raised the event.</para>
</parameter>
<parameter name="ACLName" required="false">
<para>If available, the name of the ACL that failed.</para>
</parameter>
<parameter name="SessionTV" required="false">
<para>The timestamp reported by the session.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="InvalidAccountID">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request fails an authentication check due to an invalid account ID.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SessionLimit">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request fails due to exceeding the number of allowed concurrent sessions for that service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MemoryLimit">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request fails due to an internal memory allocation failure.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="LoadAverageLimit">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request fails because a configured load average limit has been reached.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="RequestNotSupported">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request fails due to some aspect of the requested item not being supported by the service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="RequestType">
<para>The type of request attempted.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="RequestNotAllowed">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request is not allowed by the service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RequestNotSupported']/managerEventInstance/syntax/parameter[@name='RequestType'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
<parameter name="RequestParams" required="false">
<para>Parameters provided to the rejected request.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AuthMethodNotAllowed">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request used an authentication method not allowed by the service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="AuthMethod">
<para>The authentication method attempted.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="RequestBadFormat">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request is received with bad formatting.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RequestNotSupported']/managerEventInstance/syntax/parameter[@name='RequestType'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
<parameter name="AccountID" required="false">
<para>The account ID associated with the rejected request.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RequestNotAllowed']/managerEventInstance/syntax/parameter[@name='RequestParams'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SuccessfulAuth">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request successfully authenticates with a service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="UsingPassword">
<para>Whether or not the authentication attempt included a password.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="UnexpectedAddress">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request has a different source address then what is expected for a session already in progress with a service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="ExpectedAddress">
<para>The address that the request was expected to use.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChallengeResponseFailed">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request's attempt to authenticate has been challenged, and the request failed the authentication challenge.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="Challenge">
<para>The challenge that was sent.</para>
</parameter>
<parameter name="Response">
<para>The response that was received.</para>
</parameter>
<parameter name="ExpectedResponse">
<para>The expected response to the challenge.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="InvalidPassword">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request provides an invalid password during an authentication attempt.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
<parameter name="Challenge" required="false">
<para>The challenge that was sent.</para>
</parameter>
<parameter name="ReceivedChallenge" required="false">
<para>The challenge that was received.</para>
</parameter>
<parameter name="RecievedHash" required="false">
<para>The hash that was received.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChallengeSent">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when an Asterisk service sends an authentication challenge to a request.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ChallengeResponseFailed']/managerEventInstance/syntax/parameter[@name='Challenge'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="InvalidTransport">
<managerEventInstance class="EVENT_FLAG_SECURITY">
<synopsis>Raised when a request attempts to use a transport not allowed by the Asterisk service.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventTV'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Severity'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Service'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='EventVersion'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='AccountID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='LocalAddress'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='RemoteAddress'])" />
<parameter name="AttemptedTransport">
<para>The transport type that the request attempted to use.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='Module'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FailedACL']/managerEventInstance/syntax/parameter[@name='SessionTV'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<application name="Set" language="en_US">
<synopsis>
Set channel variable or function value.
</synopsis>
<syntax argsep="=">
<parameter name="name" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>This function can be used to set the value of channel variables or dialplan functions.
When setting variables, if the variable name is prefixed with <literal>_</literal>,
the variable will be inherited into channels created from the current channel.
If the variable name is prefixed with <literal>__</literal>, the variable will be
inherited into channels created from the current channel and all children channels.</para>
<note>
<para>If (and only if), in <filename>/etc/asterisk/asterisk.conf</filename>, you have
a <literal>[compat]</literal> category, and you have <literal>app_set = 1.4</literal> under that, then
the behavior of this app changes, and strips surrounding quotes from the right hand side as
it did previously in 1.4.
The advantages of not stripping out quoting, and not caring about the separator characters (comma and vertical bar)
were sufficient to make these changes in 1.6. Confusion about how many backslashes would be needed to properly
protect separators and quotes in various database access strings has been greatly
reduced by these changes.</para>
</note>
</description>
<see-also>
<ref type="application">MSet</ref>
<ref type="function">GLOBAL</ref>
<ref type="function">SET</ref>
<ref type="function">ENV</ref>
</see-also>
</application>
<application name="MSet" language="en_US">
<synopsis>
Set channel variable(s) or function value(s).
</synopsis>
<syntax>
<parameter name="set1" required="true" argsep="=">
<argument name="name1" required="true" />
<argument name="value1" required="true" />
</parameter>
<parameter name="set2" multiple="true" argsep="=">
<argument name="name2" required="true" />
<argument name="value2" required="true" />
</parameter>
</syntax>
<description>
<para>This function can be used to set the value of channel variables or dialplan functions.
When setting variables, if the variable name is prefixed with <literal>_</literal>,
the variable will be inherited into channels created from the current channel
If the variable name is prefixed with <literal>__</literal>, the variable will be
inherited into channels created from the current channel and all children channels.
MSet behaves in a similar fashion to the way Set worked in 1.2/1.4 and is thus
prone to doing things that you may not expect. For example, it strips surrounding
double-quotes from the right-hand side (value). If you need to put a separator
character (comma or vert-bar), you will need to escape them by inserting a backslash
before them. Avoid its use if possible.</para>
</description>
<see-also>
<ref type="application">Set</ref>
</see-also>
</application>
<managerEvent language="en_US" name="VarSet">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>Raised when a variable is set to a particular value.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Variable">
<para>The variable being set.</para>
</parameter>
<parameter name="Value">
<para>The new value of the variable.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentLogin">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when an Agent has logged in.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Agent">
<para>Agent ID of the agent.</para>
</parameter>
</syntax>
<see-also>
<ref type="application">AgentLogin</ref>
<ref type="managerEvent">AgentLogoff</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AgentLogoff">
<managerEventInstance class="EVENT_FLAG_AGENT">
<synopsis>Raised when an Agent has logged off.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AgentLogin']/managerEventInstance/syntax/parameter)" />
<parameter name="Logintime">
<para>The number of seconds the agent was logged in.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AgentLogin</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChannelTalkingStart">
<managerEventInstance class="EVENT_FLAG_CLASS">
<synopsis>Raised when talking is detected on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="function">TALK_DETECT</ref>
<ref type="managerEvent">ChannelTalkingStop</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChannelTalkingStop">
<managerEventInstance class="EVENT_FLAG_CLASS">
<synopsis>Raised when talking is no longer detected on a channel.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Duration">
<para>The length in time, in milliseconds, that talking was
detected on the channel.</para>
</parameter>
</syntax>
<see-also>
<ref type="function">TALK_DETECT</ref>
<ref type="managerEvent">ChannelTalkingStart</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<configInfo name="cel" language="en_US">
<configFile name="cel.conf">
<configObject name="general">
<synopsis>Options that apply globally to Channel Event Logging (CEL)</synopsis>
<configOption name="enable">
<synopsis>Determines whether CEL is enabled</synopsis>
</configOption>
<configOption name="dateformat">
<synopsis>The format to be used for dates when logging</synopsis>
</configOption>
<configOption name="apps">
<synopsis>List of apps for CEL to track</synopsis>
<description>
<para>A case-insensitive, comma-separated list of applications
to track when one or both of APP_START and APP_END events are flagged for
tracking</para>
</description>
</configOption>
<configOption name="events">
<synopsis>List of events for CEL to track</synopsis>
<description>
<para>A case-sensitive, comma-separated list of event names
to track. These event names do not include the leading <literal>AST_CEL</literal>.
</para>
<enumlist>
<enum name="ALL">
<para>Special value which tracks all events.</para>
</enum>
<enum name="CHAN_START"/>
<enum name="CHAN_END"/>
<enum name="ANSWER"/>
<enum name="HANGUP"/>
<enum name="APP_START"/>
<enum name="APP_END"/>
<enum name="PARK_START"/>
<enum name="PARK_END"/>
<enum name="USER_DEFINED"/>
<enum name="BRIDGE_ENTER"/>
<enum name="BRIDGE_EXIT"/>
<enum name="BLINDTRANSFER"/>
<enum name="ATTENDEDTRANSFER"/>
<enum name="PICKUP"/>
<enum name="FORWARD"/>
<enum name="LINKEDID_END"/>
<enum name="LOCAL_OPTIMIZE"/>
</enumlist>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="named_acl" language="en_US">
<configFile name="named_acl.conf">
<configObject name="named_acl">
<synopsis>Options for configuring a named ACL</synopsis>
<configOption name="permit">
<synopsis>An address/subnet from which to allow access</synopsis>
</configOption>
<configOption name="deny">
<synopsis>An address/subnet from which to disallow access</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<managerEvent language="en_US" name="PeerStatus">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when the state of a peer changes.</synopsis>
<syntax>
<parameter name="ChannelType">
<para>The channel technology of the peer.</para>
</parameter>
<parameter name="Peer">
<para>The name of the peer (including channel technology).</para>
</parameter>
<parameter name="PeerStatus">
<para>New status of the peer.</para>
<enumlist>
<enum name="Unknown"/>
<enum name="Registered"/>
<enum name="Unregistered"/>
<enum name="Rejected"/>
<enum name="Lagged"/>
</enumlist>
</parameter>
<parameter name="Cause">
<para>The reason the status has changed.</para>
</parameter>
<parameter name="Address">
<para>New address of the peer.</para>
</parameter>
<parameter name="Port">
<para>New port for the peer.</para>
</parameter>
<parameter name="Time">
<para>Time it takes to reach the peer and receive a response.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ContactStatus">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when the state of a contact changes.</synopsis>
<syntax>
<parameter name="URI">
<para>This contact's URI.</para>
</parameter>
<parameter name="ContactStatus">
<para>New status of the contact.</para>
<enumlist>
<enum name="Unknown"/>
<enum name="Unreachable"/>
<enum name="Reachable"/>
<enum name="Created"/>
<enum name="Removed"/>
<enum name="Updated"/>
</enumlist>
</parameter>
<parameter name="AOR">
<para>The name of the associated aor.</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the associated endpoint.</para>
</parameter>
<parameter name="RoundtripUsec">
<para>The RTT measured during the last qualify.</para>
</parameter>
<parameter name="UserAgent">
<para>Content of the User-Agent header in REGISTER request</para>
</parameter>
<parameter name="RegExpire">
<para>Absolute time that this contact is no longer valid after</para>
</parameter>
<parameter name="ViaAddress">
<para>IP address:port of the last Via header in REGISTER request</para>
</parameter>
<parameter name="CallID">
<para>Content of the Call-ID header in REGISTER request</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="FullyBooted">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when all Asterisk initialization procedures have finished.</synopsis>
<syntax>
<parameter name="Status">
<para>Informational message</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Shutdown">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when Asterisk is shutdown or restarted.</synopsis>
<syntax>
<parameter name="Shutdown">
<para>Whether the shutdown is proceeding cleanly (all channels
were hungup successfully) or uncleanly (channels will be
terminated)</para>
<enumlist>
<enum name="Uncleanly"/>
<enum name="Cleanly"/>
</enumlist>
</parameter>
<parameter name="Restart">
<para>Whether or not a restart will occur.</para>
<enumlist>
<enum name="True"/>
<enum name="False"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="RTCPSent">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>Raised when an RTCP packet is sent.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="SSRC">
<para>The SSRC identifier for our stream</para>
</parameter>
<parameter name="PT">
<para>The type of packet for this RTCP report.</para>
<enumlist>
<enum name="200(SR)"/>
<enum name="201(RR)"/>
</enumlist>
</parameter>
<parameter name="To">
<para>The address the report is sent to.</para>
</parameter>
<parameter name="ReportCount">
<para>The number of reports that were sent.</para>
<para>The report count determines the number of ReportX headers in
the message. The X for each set of report headers will range from 0 to
<literal>ReportCount - 1</literal>.</para>
</parameter>
<parameter name="SentNTP" required="false">
<para>The time the sender generated the report. Only valid when
PT is <literal>200(SR)</literal>.</para>
</parameter>
<parameter name="SentRTP" required="false">
<para>The sender's last RTP timestamp. Only valid when PT is
<literal>200(SR)</literal>.</para>
</parameter>
<parameter name="SentPackets" required="false">
<para>The number of packets the sender has sent. Only valid when PT
is <literal>200(SR)</literal>.</para>
</parameter>
<parameter name="SentOctets" required="false">
<para>The number of bytes the sender has sent. Only valid when PT is
<literal>200(SR)</literal>.</para>
</parameter>
<parameter name="ReportXSourceSSRC">
<para>The SSRC for the source of this report block.</para>
</parameter>
<parameter name="ReportXFractionLost">
<para>The fraction of RTP data packets from <literal>ReportXSourceSSRC</literal>
lost since the previous SR or RR report was sent.</para>
</parameter>
<parameter name="ReportXCumulativeLost">
<para>The total number of RTP data packets from <literal>ReportXSourceSSRC</literal>
lost since the beginning of reception.</para>
</parameter>
<parameter name="ReportXHighestSequence">
<para>The highest sequence number received in an RTP data packet from
<literal>ReportXSourceSSRC</literal>.</para>
</parameter>
<parameter name="ReportXSequenceNumberCycles">
<para>The number of sequence number cycles seen for the RTP data
received from <literal>ReportXSourceSSRC</literal>.</para>
</parameter>
<parameter name="ReportXIAJitter">
<para>An estimate of the statistical variance of the RTP data packet
interarrival time, measured in timestamp units.</para>
</parameter>
<parameter name="ReportXLSR">
<para>The last SR timestamp received from <literal>ReportXSourceSSRC</literal>.
If no SR has been received from <literal>ReportXSourceSSRC</literal>,
then 0.</para>
</parameter>
<parameter name="ReportXDLSR">
<para>The delay, expressed in units of 1/65536 seconds, between
receiving the last SR packet from <literal>ReportXSourceSSRC</literal>
and sending this report.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">RTCPReceived</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="RTCPReceived">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>Raised when an RTCP packet is received.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="SSRC">
<para>The SSRC identifier for the remote system</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='PT'])" />
<parameter name="From">
<para>The address the report was received from.</para>
</parameter>
<parameter name="RTT">
<para>Calculated Round-Trip Time in seconds</para>
</parameter>
<parameter name="ReportCount">
<para>The number of reports that were received.</para>
<para>The report count determines the number of ReportX headers in
the message. The X for each set of report headers will range from 0 to
<literal>ReportCount - 1</literal>.</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentNTP'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentRTP'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentPackets'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[@name='SentOctets'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='RTCPSent']/managerEventInstance/syntax/parameter[contains(@name, 'ReportX')])" />
</syntax>
<see-also>
<ref type="managerEvent">RTCPSent</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<application name="CallCompletionRequest" language="en_US">
<synopsis>
Request call completion service for previous call
</synopsis>
<syntax />
<description>
<para>Request call completion service for a previously failed
call attempt.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="CC_REQUEST_RESULT">
<para>This is the returned status of the request.</para>
<value name="SUCCESS" />
<value name="FAIL" />
</variable>
<variable name="CC_REQUEST_REASON">
<para>This is the reason the request failed.</para>
<value name="NO_CORE_INSTANCE" />
<value name="NOT_GENERIC" />
<value name="TOO_MANY_REQUESTS" />
<value name="UNSPECIFIED" />
</variable>
</variablelist>
</description>
</application>
<application name="CallCompletionCancel" language="en_US">
<synopsis>
Cancel call completion service
</synopsis>
<syntax />
<description>
<para>Cancel a Call Completion Request.</para>
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="CC_CANCEL_RESULT">
<para>This is the returned status of the cancel.</para>
<value name="SUCCESS" />
<value name="FAIL" />
</variable>
<variable name="CC_CANCEL_REASON">
<para>This is the reason the cancel failed.</para>
<value name="NO_CORE_INSTANCE" />
<value name="NOT_GENERIC" />
<value name="UNSPECIFIED" />
</variable>
</variablelist>
</description>
</application>
<application name="Answer" language="en_US">
<synopsis>
Answer a channel if ringing.
</synopsis>
<syntax>
<parameter name="delay">
<para>Asterisk will wait this number of milliseconds before returning to
the dialplan after answering the call.</para>
</parameter>
</syntax>
<description>
<para>If the call has not been answered, this application will
answer it. Otherwise, it has no effect on the call.</para>
</description>
<see-also>
<ref type="application">Hangup</ref>
</see-also>
</application>
<application name="BackGround" language="en_US">
<synopsis>
Play an audio file while waiting for digits of an extension to go to.
</synopsis>
<syntax>
<parameter name="filenames" required="true" argsep="&amp;">
<argument name="filename1" required="true" />
<argument name="filename2" multiple="true" />
</parameter>
<parameter name="options">
<optionlist>
<option name="s">
<para>Causes the playback of the message to be skipped
if the channel is not in the <literal>up</literal> state (i.e. it
hasn't been answered yet). If this happens, the
application will return immediately.</para>
</option>
<option name="n">
<para>Don't answer the channel before playing the files.</para>
</option>
<option name="m">
<para>Only break if a digit hit matches a one digit
extension in the destination context.</para>
</option>
</optionlist>
</parameter>
<parameter name="langoverride">
<para>Explicitly specifies which language to attempt to use for the requested sound files.</para>
</parameter>
<parameter name="context">
<para>This is the dialplan context that this application will use when exiting
to a dialed extension.</para>
</parameter>
</syntax>
<description>
<para>This application will play the given list of files <emphasis>(do not put extension)</emphasis>
while waiting for an extension to be dialed by the calling channel. To continue waiting
for digits after this application has finished playing files, the <literal>WaitExten</literal>
application should be used.</para>
<para>If one of the requested sound files does not exist, call processing will be terminated.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="BACKGROUNDSTATUS">
<para>The status of the background attempt as a text string.</para>
<value name="SUCCESS" />
<value name="FAILED" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">ControlPlayback</ref>
<ref type="application">WaitExten</ref>
<ref type="application">BackgroundDetect</ref>
<ref type="function">TIMEOUT</ref>
</see-also>
</application>
<application name="Busy" language="en_US">
<synopsis>
Indicate the Busy condition.
</synopsis>
<syntax>
<parameter name="timeout">
<para>If specified, the calling channel will be hung up after the specified number of seconds.
Otherwise, this application will wait until the calling channel hangs up.</para>
</parameter>
</syntax>
<description>
<para>This application will indicate the busy condition to the calling channel.</para>
</description>
<see-also>
<ref type="application">Congestion</ref>
<ref type="application">Progress</ref>
<ref type="application">Playtones</ref>
<ref type="application">Hangup</ref>
</see-also>
</application>
<application name="Congestion" language="en_US">
<synopsis>
Indicate the Congestion condition.
</synopsis>
<syntax>
<parameter name="timeout">
<para>If specified, the calling channel will be hung up after the specified number of seconds.
Otherwise, this application will wait until the calling channel hangs up.</para>
</parameter>
</syntax>
<description>
<para>This application will indicate the congestion condition to the calling channel.</para>
</description>
<see-also>
<ref type="application">Busy</ref>
<ref type="application">Progress</ref>
<ref type="application">Playtones</ref>
<ref type="application">Hangup</ref>
</see-also>
</application>
<application name="ExecIfTime" language="en_US">
<synopsis>
Conditional application execution based on the current time.
</synopsis>
<syntax argsep="?">
<parameter name="day_condition" required="true">
<argument name="times" required="true" />
<argument name="weekdays" required="true" />
<argument name="mdays" required="true" />
<argument name="months" required="true" />
<argument name="timezone" required="false" />
</parameter>
<parameter name="appname" required="true" hasparams="optional">
<argument name="appargs" required="true" />
</parameter>
</syntax>
<description>
<para>This application will execute the specified dialplan application, with optional
arguments, if the current time matches the given time specification.</para>
</description>
<see-also>
<ref type="application">Exec</ref>
<ref type="application">ExecIf</ref>
<ref type="application">TryExec</ref>
<ref type="application">GotoIfTime</ref>
</see-also>
</application>
<application name="Goto" language="en_US">
<synopsis>
Jump to a particular priority, extension, or context.
</synopsis>
<syntax>
<parameter name="context" />
<parameter name="extensions" />
<parameter name="priority" required="true" />
</syntax>
<description>
<para>This application will set the current context, extension, and priority in the channel structure.
After it completes, the pbx engine will continue dialplan execution at the specified location.
If no specific <replaceable>extension</replaceable>, or <replaceable>extension</replaceable> and
<replaceable>context</replaceable>, are specified, then this application will
just set the specified <replaceable>priority</replaceable> of the current extension.</para>
<para>At least a <replaceable>priority</replaceable> is required as an argument, or the goto will
return a <literal>-1</literal>, and the channel and call will be terminated.</para>
<para>If the location that is put into the channel information is bogus, and asterisk cannot
find that location in the dialplan, then the execution engine will try to find and execute the code in
the <literal>i</literal> (invalid) extension in the current context. If that does not exist, it will try to execute the
<literal>h</literal> extension. If neither the <literal>h</literal> nor <literal>i</literal> extensions
have been defined, the channel is hung up, and the execution of instructions on the channel is terminated.
What this means is that, for example, you specify a context that does not exist, then
it will not be possible to find the <literal>h</literal> or <literal>i</literal> extensions,
and the call will terminate!</para>
</description>
<see-also>
<ref type="application">GotoIf</ref>
<ref type="application">GotoIfTime</ref>
<ref type="application">Gosub</ref>
<ref type="application">Macro</ref>
</see-also>
</application>
<application name="GotoIf" language="en_US">
<synopsis>
Conditional goto.
</synopsis>
<syntax argsep="?">
<parameter name="condition" required="true" />
<parameter name="destination" required="true" argsep=":">
<argument name="labeliftrue">
<para>Continue at <replaceable>labeliftrue</replaceable> if the condition is true.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
</argument>
<argument name="labeliffalse">
<para>Continue at <replaceable>labeliffalse</replaceable> if the condition is false.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
</argument>
</parameter>
</syntax>
<description>
<para>This application will set the current context, extension, and priority in the channel structure
based on the evaluation of the given condition. After this application completes, the
pbx engine will continue dialplan execution at the specified location in the dialplan.
The labels are specified with the same syntax as used within the Goto application.
If the label chosen by the condition is omitted, no jump is performed, and the execution passes to the
next instruction. If the target location is bogus, and does not exist, the execution engine will try
to find and execute the code in the <literal>i</literal> (invalid) extension in the current context.
If that does not exist, it will try to execute the <literal>h</literal> extension.
If neither the <literal>h</literal> nor <literal>i</literal> extensions have been defined,
the channel is hung up, and the execution of instructions on the channel is terminated.
Remember that this command can set the current context, and if the context specified
does not exist, then it will not be able to find any 'h' or 'i' extensions there, and
the channel and call will both be terminated!.</para>
</description>
<see-also>
<ref type="application">Goto</ref>
<ref type="application">GotoIfTime</ref>
<ref type="application">GosubIf</ref>
<ref type="application">MacroIf</ref>
</see-also>
</application>
<application name="GotoIfTime" language="en_US">
<synopsis>
Conditional Goto based on the current time.
</synopsis>
<syntax argsep="?">
<parameter name="condition" required="true">
<argument name="times" required="true" />
<argument name="weekdays" required="true" />
<argument name="mdays" required="true" />
<argument name="months" required="true" />
<argument name="timezone" required="false" />
</parameter>
<parameter name="destination" required="true" argsep=":">
<argument name="labeliftrue">
<para>Continue at <replaceable>labeliftrue</replaceable> if the condition is true.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
</argument>
<argument name="labeliffalse">
<para>Continue at <replaceable>labeliffalse</replaceable> if the condition is false.
Takes the form similar to Goto() of [[context,]extension,]priority.</para>
</argument>
</parameter>
</syntax>
<description>
<para>This application will set the context, extension, and priority in the channel structure
based on the evaluation of the given time specification. After this application completes,
the pbx engine will continue dialplan execution at the specified location in the dialplan.
If the current time is within the given time specification, the channel will continue at
<replaceable>labeliftrue</replaceable>. Otherwise the channel will continue at <replaceable>labeliffalse</replaceable>.
If the label chosen by the condition is omitted, no jump is performed, and execution passes to the next
instruction. If the target jump location is bogus, the same actions would be taken as for <literal>Goto</literal>.
Further information on the time specification can be found in examples
illustrating how to do time-based context includes in the dialplan.</para>
</description>
<see-also>
<ref type="application">GotoIf</ref>
<ref type="application">Goto</ref>
<ref type="function">IFTIME</ref>
<ref type="function">TESTTIME</ref>
</see-also>
</application>
<application name="ImportVar" language="en_US">
<synopsis>
Import a variable from a channel into a new variable.
</synopsis>
<syntax argsep="=">
<parameter name="newvar" required="true" />
<parameter name="vardata" required="true">
<argument name="channelname" required="true" />
<argument name="variable" required="true" />
</parameter>
</syntax>
<description>
<para>This application imports a <replaceable>variable</replaceable> from the specified
<replaceable>channel</replaceable> (as opposed to the current one) and stores it as a variable
(<replaceable>newvar</replaceable>) in the current channel (the channel that is calling this
application). Variables created by this application have the same inheritance properties as those
created with the <literal>Set</literal> application.</para>
</description>
<see-also>
<ref type="application">Set</ref>
</see-also>
</application>
<application name="Hangup" language="en_US">
<synopsis>
Hang up the calling channel.
</synopsis>
<syntax>
<parameter name="causecode">
<para>If a <replaceable>causecode</replaceable> is given the channel's
hangup cause will be set to the given value.</para>
</parameter>
</syntax>
<description>
<para>This application will hang up the calling channel.</para>
</description>
<see-also>
<ref type="application">Answer</ref>
<ref type="application">Busy</ref>
<ref type="application">Congestion</ref>
</see-also>
</application>
<application name="Incomplete" language="en_US">
<synopsis>
Returns AST_PBX_INCOMPLETE value.
</synopsis>
<syntax>
<parameter name="n">
<para>If specified, then Incomplete will not attempt to answer the channel first.</para>
<note>
<para>Most channel types need to be in Answer state in order to receive DTMF.</para>
</note>
</parameter>
</syntax>
<description>
<para>Signals the PBX routines that the previous matched extension is incomplete
and that further input should be allowed before matching can be considered
to be complete. Can be used within a pattern match when certain criteria warrants
a longer match.</para>
</description>
</application>
<application name="NoOp" language="en_US">
<synopsis>
Do Nothing (No Operation).
</synopsis>
<syntax>
<parameter name="text">
<para>Any text provided can be viewed at the Asterisk CLI.</para>
</parameter>
</syntax>
<description>
<para>This application does nothing. However, it is useful for debugging purposes.</para>
<para>This method can be used to see the evaluations of variables or functions without having any effect.</para>
</description>
<see-also>
<ref type="application">Verbose</ref>
<ref type="application">Log</ref>
</see-also>
</application>
<application name="Proceeding" language="en_US">
<synopsis>
Indicate proceeding.
</synopsis>
<syntax />
<description>
<para>This application will request that a proceeding message be provided to the calling channel.</para>
</description>
</application>
<application name="Progress" language="en_US">
<synopsis>
Indicate progress.
</synopsis>
<syntax />
<description>
<para>This application will request that in-band progress information be provided to the calling channel.</para>
</description>
<see-also>
<ref type="application">Busy</ref>
<ref type="application">Congestion</ref>
<ref type="application">Ringing</ref>
<ref type="application">Playtones</ref>
</see-also>
</application>
<application name="RaiseException" language="en_US">
<synopsis>
Handle an exceptional condition.
</synopsis>
<syntax>
<parameter name="reason" required="true" />
</syntax>
<description>
<para>This application will jump to the <literal>e</literal> extension in the current context, setting the
dialplan function EXCEPTION(). If the <literal>e</literal> extension does not exist, the call will hangup.</para>
</description>
<see-also>
<ref type="function">Exception</ref>
</see-also>
</application>
<application name="Ringing" language="en_US">
<synopsis>
Indicate ringing tone.
</synopsis>
<syntax />
<description>
<para>This application will request that the channel indicate a ringing tone to the user.</para>
</description>
<see-also>
<ref type="application">Busy</ref>
<ref type="application">Congestion</ref>
<ref type="application">Progress</ref>
<ref type="application">Playtones</ref>
</see-also>
</application>
<application name="SayAlpha" language="en_US">
<synopsis>
Say Alpha.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>This application will play the sounds that correspond to the letters
of the given <replaceable>string</replaceable>. If the channel variable
<variable>SAY_DTMF_INTERRUPT</variable> is set to 'true' (case insensitive),
then this application will react to DTMF in the same way as
<literal>Background</literal>.</para>
</description>
<see-also>
<ref type="application">SayDigits</ref>
<ref type="application">SayNumber</ref>
<ref type="application">SayPhonetic</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</application>
<application name="SayAlphaCase" language="en_US">
<synopsis>
Say Alpha.
</synopsis>
<syntax>
<parameter name="casetype" required="true" >
<enumlist>
<enum name="a">
<para>Case sensitive (all) pronunciation.
(Ex: SayAlphaCase(a,aBc); - lowercase a uppercase b lowercase c).</para>
</enum>
<enum name="l">
<para>Case sensitive (lower) pronunciation.
(Ex: SayAlphaCase(l,aBc); - lowercase a b lowercase c).</para>
</enum>
<enum name="n">
<para>Case insensitive pronunciation. Equivalent to SayAlpha.
(Ex: SayAlphaCase(n,aBc) - a b c).</para>
</enum>
<enum name="u">
<para>Case sensitive (upper) pronunciation.
(Ex: SayAlphaCase(u,aBc); - a uppercase b c).</para>
</enum>
</enumlist>
</parameter>
<parameter name="string" required="true" />
</syntax>
<description>
<para>This application will play the sounds that correspond to the letters of the
given <replaceable>string</replaceable>. Optionally, a <replaceable>casetype</replaceable> may be
specified. This will be used for case-insensitive or case-sensitive pronunciations. If the channel
variable <variable>SAY_DTMF_INTERRUPT</variable> is set to 'true' (case insensitive), then this
application will react to DTMF in the same way as <literal>Background</literal>.</para>
</description>
<see-also>
<ref type="application">SayDigits</ref>
<ref type="application">SayNumber</ref>
<ref type="application">SayPhonetic</ref>
<ref type="application">SayAlpha</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</application>
<application name="SayDigits" language="en_US">
<synopsis>
Say Digits.
</synopsis>
<syntax>
<parameter name="digits" required="true" />
</syntax>
<description>
<para>This application will play the sounds that correspond to the digits of
the given number. This will use the language that is currently set for the channel.
If the channel variable <variable>SAY_DTMF_INTERRUPT</variable> is set to 'true'
(case insensitive), then this application will react to DTMF in the same way as
<literal>Background</literal>.</para>
</description>
<see-also>
<ref type="application">SayAlpha</ref>
<ref type="application">SayNumber</ref>
<ref type="application">SayPhonetic</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</application>
<application name="SayNumber" language="en_US">
<synopsis>
Say Number.
</synopsis>
<syntax>
<parameter name="digits" required="true" />
<parameter name="gender" />
</syntax>
<description>
<para>This application will play the sounds that correspond to the given
<replaceable>digits</replaceable>. Optionally, a <replaceable>gender</replaceable> may be
specified. This will use the language that is currently set for the channel. See the CHANNEL()
function for more information on setting the language for the channel. If the channel variable
<variable>SAY_DTMF_INTERRUPT</variable> is set to 'true' (case insensitive), then this
application will react to DTMF in the same way as <literal>Background</literal>.</para>
</description>
<see-also>
<ref type="application">SayAlpha</ref>
<ref type="application">SayDigits</ref>
<ref type="application">SayPhonetic</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</application>
<application name="SayPhonetic" language="en_US">
<synopsis>
Say Phonetic.
</synopsis>
<syntax>
<parameter name="string" required="true" />
</syntax>
<description>
<para>This application will play the sounds from the phonetic alphabet that correspond to the
letters in the given <replaceable>string</replaceable>. If the channel variable
<variable>SAY_DTMF_INTERRUPT</variable> is set to 'true' (case insensitive), then this
application will react to DTMF in the same way as <literal>Background</literal>.</para>
</description>
<see-also>
<ref type="application">SayAlpha</ref>
<ref type="application">SayDigits</ref>
<ref type="application">SayNumber</ref>
</see-also>
</application>
<application name="SetAMAFlags" language="en_US">
<synopsis>
Set the AMA Flags.
</synopsis>
<syntax>
<parameter name="flag" />
</syntax>
<description>
<para>This application will set the channel's AMA Flags for billing purposes.</para>
<warning>
<para>This application is deprecated. Please use the CHANNEL function instead.</para>
</warning>
</description>
<see-also>
<ref type="function">CDR</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</application>
<application name="Wait" language="en_US">
<synopsis>
Waits for some time.
</synopsis>
<syntax>
<parameter name="seconds" required="true">
<para>Can be passed with fractions of a second. For example, <literal>1.5</literal> will ask the
application to wait for 1.5 seconds.</para>
</parameter>
</syntax>
<description>
<para>This application waits for a specified number of <replaceable>seconds</replaceable>.</para>
</description>
</application>
<application name="WaitExten" language="en_US">
<synopsis>
Waits for an extension to be entered.
</synopsis>
<syntax>
<parameter name="seconds">
<para>Can be passed with fractions of a second. For example, <literal>1.5</literal> will ask the
application to wait for 1.5 seconds.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="m">
<para>Provide music on hold to the caller while waiting for an extension.</para>
<argument name="x">
<para>Specify the class for music on hold. <emphasis>CHANNEL(musicclass) will
be used instead if set</emphasis>
</para>
</argument>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application waits for the user to enter a new extension for a specified number
of <replaceable>seconds</replaceable>.</para>
<xi:include xpointer="xpointer(/docs/application[@name='Macro']/description/warning[2])" />
</description>
<see-also>
<ref type="application">Background</ref>
<ref type="function">TIMEOUT</ref>
</see-also>
</application>
<managerEvent language="en_US" name="Reload">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when a module has been reloaded in Asterisk.</synopsis>
<syntax>
<parameter name="Module">
<para>The name of the module that was reloaded, or
<literal>All</literal> if all modules were reloaded</para>
</parameter>
<parameter name="Status">
<para>The numeric status code denoting the success or failure
of the reload request.</para>
<enumlist>
<enum name="0">
<para>Success</para>
</enum>
<enum name="1">
<para>Request queued</para>
</enum>
<enum name="2">
<para>Module not found</para>
</enum>
<enum name="3">
<para>Error</para>
</enum>
<enum name="4">
<para>Reload already in progress</para>
</enum>
<enum name="5">
<para>Module uninitialized</para>
</enum>
<enum name="6">
<para>Reload not supported</para>
</enum>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="PresenceStateChange">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a presence state changes</synopsis>
<syntax>
<parameter name="Presentity">
<para>The entity whose presence state has changed</para>
</parameter>
<parameter name="Status">
<para>The new status of the presentity</para>
</parameter>
<parameter name="Subtype">
<para>The new subtype of the presentity</para>
</parameter>
<parameter name="Message">
<para>The new message of the presentity</para>
</parameter>
</syntax>
<description>
<para>This differs from the <literal>PresenceStatus</literal>
event because this event is raised for all presence state changes,
not only for changes that affect dialplan hints.</para>
</description>
<see-also>
<ref type="managerEvent">PresenceStatus</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="DeviceStateChange">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a device state changes</synopsis>
<syntax>
<parameter name="Device">
<para>The device whose state has changed</para>
</parameter>
<parameter name="State">
<para>The new state of the device</para>
</parameter>
</syntax>
<description>
<para>This differs from the <literal>ExtensionStatus</literal>
event because this event is raised for all device state changes,
not only for changes that affect dialplan hints.</para>
</description>
<see-also>
<ref type="managerEvent">ExtensionStatus</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Newchannel">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a new channel is created.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">Newstate</ref>
<ref type="managerEvent">Hangup</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Newstate">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel's state changes.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">Newchannel</ref>
<ref type="managerEvent">Hangup</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Hangup">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel is hung up.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Cause">
<para>A numeric cause code for why the channel was hung up.</para>
</parameter>
<parameter name="Cause-txt">
<para>A description of why the channel was hung up.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">Newchannel</ref>
<ref type="managerEvent">SoftHangupRequest</ref>
<ref type="managerEvent">HangupRequest</ref>
<ref type="managerEvent">Newstate</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="HangupRequest">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a hangup is requested.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='Hangup']/managerEventInstance/syntax/parameter[@name='Cause'])" />
</syntax>
<see-also>
<ref type="managerEvent">SoftHangupRequest</ref>
<ref type="managerEvent">Hangup</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SoftHangupRequest">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a soft hangup is requested with a specific cause code.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='Hangup']/managerEventInstance/syntax/parameter[@name='Cause'])" />
</syntax>
<see-also>
<ref type="managerEvent">HangupRequest</ref>
<ref type="managerEvent">Hangup</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="NewExten">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>Raised when a channel enters a new context, extension, priority.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Extension">
<para>Deprecated in 12, but kept for
backward compatability. Please use
'Exten' instead.</para>
</parameter>
<parameter name="Application">
<para>The application about to be executed.</para>
</parameter>
<parameter name="AppData">
<para>The data to be passed to the application.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="NewCallerid">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel receives new Caller ID information.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="CID-CallingPres">
<para>A description of the Caller ID presentation.</para>
</parameter>
</syntax>
<see-also>
<ref type="function">CALLERID</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="NewConnectedLine">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel's connected line information is changed.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="function">CONNECTEDLINE</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="NewAccountCode">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a Channel's AccountCode is changed.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="OldAccountCode">
<para>The channel's previous account code</para>
</parameter>
</syntax>
<see-also>
<ref type="function">CHANNEL</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="DialBegin">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a dial action has started.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<parameter name="DialString">
<para>The non-technology specific device being dialed.</para>
</parameter>
</syntax>
<see-also>
<ref type="application">Dial</ref>
<ref type="application">Originate</ref>
<ref type="manager">Originate</ref>
<ref type="managerEvent">DialEnd</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="DialEnd">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a dial action has completed.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Dest"/>
<parameter name="DialStatus">
<para>The result of the dial operation.</para>
<enumlist>
<enum name="ABORT">
<para>The call was aborted.</para>
</enum>
<enum name="ANSWER">
<para>The caller answered.</para>
</enum>
<enum name="BUSY">
<para>The caller was busy.</para>
</enum>
<enum name="CANCEL">
<para>The caller cancelled the call.</para>
</enum>
<enum name="CHANUNAVAIL">
<para>The requested channel is unavailable.</para>
</enum>
<enum name="CONGESTION">
<para>The called party is congested.</para>
</enum>
<enum name="CONTINUE">
<para>The dial completed, but the caller elected
to continue in the dialplan.</para>
</enum>
<enum name="GOTO">
<para>The dial completed, but the caller jumped to
a dialplan location.</para>
<para>If known, the location the caller is jumping
to will be appended to the result following a
":".</para>
</enum>
<enum name="NOANSWER">
<para>The called party failed to answer.</para>
</enum>
</enumlist>
</parameter>
<parameter name="Forward" required="false">
<para>If the call was forwarded, where the call was
forwarded to.</para>
</parameter>
</syntax>
<see-also>
<ref type="application">Dial</ref>
<ref type="application">Originate</ref>
<ref type="manager">Originate</ref>
<ref type="managerEvent">DialBegin</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Hold">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel goes on hold.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="MusicClass">
<para>The suggested MusicClass, if provided.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">Unhold</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Unhold">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel goes off hold.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">Hold</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChanSpyStart">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when one channel begins spying on another channel.</synopsis>
<syntax>
<channel_snapshot prefix="Spyer"/>
<channel_snapshot prefix="Spyee"/>
</syntax>
<see-also>
<ref type="managerEvent">ChanSpyStop</ref>
<ref type="application">ChanSpy</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ChanSpyStop">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel has stopped spying.</synopsis>
<syntax>
<channel_snapshot prefix="Spyer"/>
<channel_snapshot prefix="Spyee"/>
</syntax>
<see-also>
<ref type="managerEvent">ChanSpyStart</ref>
<ref type="application">ChanSpy</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="HangupHandlerRun">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>Raised when a hangup handler is about to be called.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Handler">
<para>Hangup handler parameter string passed to the Gosub application.</para>
</parameter>
</syntax>
<see-also>
<ref type="function">CHANNEL</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="HangupHandlerPop">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>
Raised when a hangup handler is removed from the handler stack
by the CHANNEL() function.
</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='HangupHandlerRun']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="managerEvent">HangupHandlerPush</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="HangupHandlerPush">
<managerEventInstance class="EVENT_FLAG_DIALPLAN">
<synopsis>
Raised when a hangup handler is added to the handler stack by
the CHANNEL() function.
</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='HangupHandlerRun']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="managerEvent">HangupHandlerPop</ref>
<ref type="function">CHANNEL</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="FAXStatus">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>
Raised periodically during a fax transmission.
</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Operation">
<enumlist>
<enum name="gateway"/>
<enum name="receive"/>
<enum name="send"/>
</enumlist>
</parameter>
<parameter name="Status">
<para>A text message describing the current status of the fax</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ReceiveFAX']/managerEventInstance/syntax/parameter[@name='LocalStationID'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ReceiveFAX']/managerEventInstance/syntax/parameter[@name='FileName'])" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ReceiveFAX">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>
Raised when a receive fax operation has completed.
</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="LocalStationID">
<para>The value of the <variable>LOCALSTATIONID</variable> channel variable</para>
</parameter>
<parameter name="RemoteStationID">
<para>The value of the <variable>REMOTESTATIONID</variable> channel variable</para>
</parameter>
<parameter name="PagesTransferred">
<para>The number of pages that have been transferred</para>
</parameter>
<parameter name="Resolution">
<para>The negotiated resolution</para>
</parameter>
<parameter name="TransferRate">
<para>The negotiated transfer rate</para>
</parameter>
<parameter name="FileName" multiple="yes">
<para>The files being affected by the fax operation</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="SendFAX">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>
Raised when a send fax operation has completed.
</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ReceiveFAX']/managerEventInstance/syntax/parameter)" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MusicOnHoldStart">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when music on hold has started on a channel.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Class">
<para>The class of music being played on the channel</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">MusicOnHoldStop</ref>
<ref type="application">StartMusicOnHold</ref>
<ref type="application">MusicOnHold</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MusicOnHoldStop">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when music on hold has stopped on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MusicOnHoldStart</ref>
<ref type="application">StopMusicOnHold</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MonitorStart">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when monitoring has started on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MonitorStop</ref>
<ref type="application">Monitor</ref>
<ref type="manager">Monitor</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MonitorStop">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when monitoring has stopped on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MonitorStart</ref>
<ref type="application">StopMonitor</ref>
<ref type="manager">StopMonitor</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<function name="EXCEPTION" language="en_US">
<synopsis>
Retrieve the details of the current dialplan exception.
</synopsis>
<syntax>
<parameter name="field" required="true">
<para>The following fields are available for retrieval:</para>
<enumlist>
<enum name="reason">
<para>INVALID, ERROR, RESPONSETIMEOUT, ABSOLUTETIMEOUT, or custom
value set by the RaiseException() application</para>
</enum>
<enum name="context">
<para>The context executing when the exception occurred.</para>
</enum>
<enum name="exten">
<para>The extension executing when the exception occurred.</para>
</enum>
<enum name="priority">
<para>The numeric priority executing when the exception occurred.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Retrieve the details (specified <replaceable>field</replaceable>) of the current dialplan exception.</para>
</description>
<see-also>
<ref type="application">RaiseException</ref>
</see-also>
</function>
<function name="TESTTIME" language="en_US">
<synopsis>
Sets a time to be used with the channel to test logical conditions.
</synopsis>
<syntax>
<parameter name="date" required="true" argsep=" ">
<para>Date in ISO 8601 format</para>
</parameter>
<parameter name="time" required="true" argsep=" ">
<para>Time in HH:MM:SS format (24-hour time)</para>
</parameter>
<parameter name="zone" required="false">
<para>Timezone name</para>
</parameter>
</syntax>
<description>
<para>To test dialplan timing conditions at times other than the current time, use
this function to set an alternate date and time. For example, you may wish to evaluate
whether a location will correctly identify to callers that the area is closed on Christmas
Day, when Christmas would otherwise fall on a day when the office is normally open.</para>
</description>
<see-also>
<ref type="application">GotoIfTime</ref>
</see-also>
</function>
<manager name="ShowDialPlan" language="en_US">
<synopsis>
Show dialplan contexts and extensions
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Extension">
<para>Show a specific extension.</para>
</parameter>
<parameter name="Context">
<para>Show a specific context.</para>
</parameter>
</syntax>
<description>
<para>Show dialplan contexts and extensions. Be aware that showing the full dialplan
may take a lot of capacity.</para>
</description>
</manager>
<manager name="ExtensionStateList" language="en_US">
<synopsis>
List the current known extension states.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>This will list out all known extension states in a
sequence of <replaceable>ExtensionStatus</replaceable> events.
When finished, a <replaceable>ExtensionStateListComplete</replaceable> event
will be emitted.</para>
</description>
<see-also>
<ref type="manager">ExtensionState</ref>
<ref type="function">HINT</ref>
<ref type="function">EXTENSION_STATE</ref>
</see-also>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ExtensionStatus'])" />
</list-elements>
<managerEvent name="ExtensionStateListComplete" language="en_US">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>
Indicates the end of the list the current known extension states.
</synopsis>
<syntax>
<parameter name="EventList">
<para>Conveys the status of the event list.</para>
</parameter>
<parameter name="ListItems">
<para>Conveys the number of statuses reported.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<managerEvent language="en_US" name="Pickup">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a call pickup occurs.</synopsis>
<syntax>
<channel_snapshot/>
<channel_snapshot prefix="Target"/>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="Ping" language="en_US">
<synopsis>
Keepalive command.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>A 'Ping' action will ellicit a 'Pong' response. Used to keep the
manager connection open.</para>
</description>
</manager>
<manager name="Events" language="en_US">
<synopsis>
Control Event Flow.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="EventMask" required="true">
<enumlist>
<enum name="on">
<para>If all events should be sent.</para>
</enum>
<enum name="off">
<para>If no events should be sent.</para>
</enum>
<enum name="system,call,log,...">
<para>To select which flags events should have to be sent.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Enable/Disable sending of events to this manager client.</para>
</description>
</manager>
<manager name="Logoff" language="en_US">
<synopsis>
Logoff Manager.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Logoff the current manager session.</para>
</description>
<see-also>
<ref type="manager">Login</ref>
</see-also>
</manager>
<manager name="Login" language="en_US">
<synopsis>
Login Manager.
</synopsis>
<syntax>
<parameter name="ActionID">
<para>ActionID for this transaction. Will be returned.</para>
</parameter>
<parameter name="Username" required="true">
<para>Username to login with as specified in manager.conf.</para>
</parameter>
<parameter name="Secret">
<para>Secret to login with as specified in manager.conf.</para>
</parameter>
</syntax>
<description>
<para>Login Manager.</para>
</description>
<see-also>
<ref type="manager">Logoff</ref>
</see-also>
</manager>
<manager name="Challenge" language="en_US">
<synopsis>
Generate Challenge for MD5 Auth.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="AuthType" required="true">
<para>Digest algorithm to use in the challenge. Valid values are:</para>
<enumlist>
<enum name="MD5" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Generate a challenge for MD5 authentication.</para>
</description>
</manager>
<manager name="Hangup" language="en_US">
<synopsis>
Hangup channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The exact channel name to be hungup, or to use a regular expression, set this parameter to: /regex/</para>
<para>Example exact channel: SIP/provider-0000012a</para>
<para>Example regular expression: /^SIP/provider-.*$/</para>
</parameter>
<parameter name="Cause">
<para>Numeric hangup cause.</para>
</parameter>
</syntax>
<description>
<para>Hangup a channel.</para>
</description>
</manager>
<manager name="Status" language="en_US">
<synopsis>
List channel status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="false">
<para>The name of the channel to query for status.</para>
</parameter>
<parameter name="Variables">
<para>Comma <literal>,</literal> separated list of variable to include.</para>
</parameter>
</syntax>
<description>
<para>Will return the status information of each channel along with the
value for the specified channel variables.</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='Status'])" />
</list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='StatusComplete'])" />
</responses>
</manager>
<managerEvent language="en_US" name="Status">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a Status command.</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<channel_snapshot/>
<parameter name="Type">
<para>Type of channel</para>
</parameter>
<parameter name="DNID">
<para>Dialed number identifier</para>
</parameter>
<parameter name="TimeToHangup">
<para>Absolute lifetime of the channel</para>
</parameter>
<parameter name="BridgeID">
<para>Identifier of the bridge the channel is in, may be empty if not in one</para>
</parameter>
<parameter name="Linkedid">
</parameter>
<parameter name="Application">
<para>Application currently executing on the channel</para>
</parameter>
<parameter name="Data">
<para>Data given to the currently executing channel</para>
</parameter>
<parameter name="Nativeformats">
<para>Media formats the connected party is willing to send or receive</para>
</parameter>
<parameter name="Readformat">
<para>Media formats that frames from the channel are received in</para>
</parameter>
<parameter name="Readtrans">
<para>Translation path for media received in native formats</para>
</parameter>
<parameter name="Writeformat">
<para>Media formats that frames to the channel are accepted in</para>
</parameter>
<parameter name="Writetrans">
<para>Translation path for media sent to the connected party</para>
</parameter>
<parameter name="Callgroup">
<para>Configured call group on the channel</para>
</parameter>
<parameter name="Pickupgroup">
<para>Configured pickup group on the channel</para>
</parameter>
<parameter name="Seconds">
<para>Number of seconds the channel has been active</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">Status</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="StatusComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a Status command.</synopsis>
<syntax>
<parameter name="Items">
<para>Number of Status events returned</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">Status</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="Setvar" language="en_US">
<synopsis>
Sets a channel variable or function value.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel">
<para>Channel to set variable for.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Variable name, function or expression.</para>
</parameter>
<parameter name="Value" required="true">
<para>Variable or function value.</para>
</parameter>
</syntax>
<description>
<para>This command can be used to set the value of channel variables or dialplan
functions.</para>
<note>
<para>If a channel name is not provided then the variable is considered global.</para>
</note>
</description>
<see-also>
<ref type="manager">Getvar</ref>
</see-also>
</manager>
<manager name="Getvar" language="en_US">
<synopsis>
Gets a channel variable or function value.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel">
<para>Channel to read variable from.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Variable name, function or expression.</para>
</parameter>
</syntax>
<description>
<para>Get the value of a channel variable or function return.</para>
<note>
<para>If a channel name is not provided then the variable is considered global.</para>
</note>
</description>
<see-also>
<ref type="manager">Setvar</ref>
</see-also>
</manager>
<manager name="GetConfig" language="en_US">
<synopsis>
Retrieve configuration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
<parameter name="Category">
<para>Category in configuration file.</para>
</parameter>
<parameter name="Filter">
<para>A comma separated list of
<replaceable>name_regex</replaceable>=<replaceable>value_regex</replaceable>
expressions which will cause only categories whose variables match all expressions
to be considered. The special variable name <literal>TEMPLATES</literal>
can be used to control whether templates are included. Passing
<literal>include</literal> as the value will include templates
along with normal categories. Passing
<literal>restrict</literal> as the value will restrict the operation to
ONLY templates. Not specifying a <literal>TEMPLATES</literal> expression
results in the default behavior which is to not include templates.</para>
</parameter>
</syntax>
<description>
<para>This action will dump the contents of a configuration
file by category and contents or optionally by specified category only.
In the case where a category name is non-unique, a filter may be specified
to match only categories with matching variable values.</para>
</description>
<see-also>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="GetConfigJSON" language="en_US">
<synopsis>
Retrieve configuration (JSON format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
<parameter name="Category">
<para>Category in configuration file.</para>
</parameter>
<parameter name="Filter">
<xi:include xpointer="xpointer(/docs/manager[@name='GetConfig']/syntax/parameter[@name='Filter']/para[1])" />
</parameter>
</syntax>
<description>
<para>This action will dump the contents of a configuration file by category
and contents in JSON format or optionally by specified category only.
This only makes sense to be used using rawman over the HTTP interface.
In the case where a category name is non-unique, a filter may be specified
to match only categories with matching variable values.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="UpdateConfig" language="en_US">
<synopsis>
Update basic configuration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="SrcFilename" required="true">
<para>Configuration filename to read (e.g. <filename>foo.conf</filename>).</para>
</parameter>
<parameter name="DstFilename" required="true">
<para>Configuration filename to write (e.g. <filename>foo.conf</filename>)</para>
</parameter>
<parameter name="Reload">
<para>Whether or not a reload should take place (or name of specific module).</para>
</parameter>
<parameter name="PreserveEffectiveContext">
<para>Whether the effective category contents should be preserved on template change. Default is true (pre 13.2 behavior).</para>
</parameter>
<parameter name="Action-000000">
<para>Action to take.</para>
<para>0's represent 6 digit number beginning with 000000.</para>
<enumlist>
<enum name="NewCat" />
<enum name="RenameCat" />
<enum name="DelCat" />
<enum name="EmptyCat" />
<enum name="Update" />
<enum name="Delete" />
<enum name="Append" />
<enum name="Insert" />
</enumlist>
</parameter>
<parameter name="Cat-000000">
<para>Category to operate on.</para>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Var-000000">
<para>Variable to work on.</para>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Value-000000">
<para>Value to work on.</para>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Match-000000">
<para>Extra match required to match line.</para>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Line-000000">
<para>Line in category to operate on (used with delete and insert actions).</para>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Options-000000">
<para>A comma separated list of action-specific options.</para>
<enumlist>
<enum name="NewCat">
<para>One or more of the following... </para>
<enumlist>
<enum name="allowdups">
<para>Allow duplicate category names.</para>
</enum>
<enum name="template">
<para>This category is a template.</para>
</enum>
<enum name="inherit=&quot;template[,...]&quot;">
<para>Templates from which to inherit.</para>
</enum>
</enumlist>
</enum>
</enumlist>
<para> </para>
<para>The following actions share the same options...</para>
<enumlist>
<enum name="RenameCat"/>
<enum name="DelCat"/>
<enum name="EmptyCat"/>
<enum name="Update"/>
<enum name="Delete"/>
<enum name="Append"/>
<enum name="Insert">
<para> </para>
<enumlist>
<enum name="catfilter=&quot;&lt;expression&gt;[,...]&quot;">
<para> </para>
<xi:include xpointer="xpointer(/docs/manager[@name='GetConfig']/syntax/parameter[@name='Filter']/para[1])" />
<para>
<literal>catfilter</literal> is most useful when a file
contains multiple categories with the same name and you wish to
operate on specific ones instead of all of them.</para>
</enum>
</enumlist>
</enum>
</enumlist>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
</syntax>
<description>
<para>This action will modify, create, or delete configuration elements
in Asterisk configuration files.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="CreateConfig" language="en_US">
<synopsis>
Creates an empty file in the configuration directory.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>The configuration filename to create (e.g. <filename>foo.conf</filename>).</para>
</parameter>
</syntax>
<description>
<para>This action will create an empty file in the configuration
directory. This action is intended to be used before an UpdateConfig
action.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="ListCategories" language="en_US">
<synopsis>
List categories in configuration file.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
</syntax>
<description>
<para>This action will dump the categories in a given file.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
</see-also>
</manager>
<manager name="Redirect" language="en_US">
<synopsis>
Redirect (transfer) a call.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel to redirect.</para>
</parameter>
<parameter name="ExtraChannel">
<para>Second call leg to transfer (optional).</para>
</parameter>
<parameter name="Exten" required="true">
<para>Extension to transfer to.</para>
</parameter>
<parameter name="ExtraExten">
<para>Extension to transfer extrachannel to (optional).</para>
</parameter>
<parameter name="Context" required="true">
<para>Context to transfer to.</para>
</parameter>
<parameter name="ExtraContext">
<para>Context to transfer extrachannel to (optional).</para>
</parameter>
<parameter name="Priority" required="true">
<para>Priority to transfer to.</para>
</parameter>
<parameter name="ExtraPriority">
<para>Priority to transfer extrachannel to (optional).</para>
</parameter>
</syntax>
<description>
<para>Redirect (transfer) a call.</para>
</description>
<see-also>
<ref type="manager">BlindTransfer</ref>
</see-also>
</manager>
<manager name="Atxfer" language="en_US">
<synopsis>
Attended transfer.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Transferer's channel.</para>
</parameter>
<parameter name="Exten" required="true">
<para>Extension to transfer to.</para>
</parameter>
<parameter name="Context">
<para>Context to transfer to.</para>
</parameter>
</syntax>
<description>
<para>Attended transfer.</para>
</description>
<see-also>
<ref type="managerEvent">AttendedTransfer</ref>
</see-also>
</manager>
<manager name="CancelAtxfer" language="en_US">
<synopsis>
Cancel an attended transfer.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The transferer channel.</para>
</parameter>
</syntax>
<description>
<para>Cancel an attended transfer. Note, this uses the configured cancel attended transfer
feature option (atxferabort) to cancel the transfer. If not available this action will fail.
</para>
</description>
<see-also>
<ref type="managerEvent">AttendedTransfer</ref>
</see-also>
</manager>
<manager name="Originate" language="en_US">
<synopsis>
Originate a call.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to call.</para>
</parameter>
<parameter name="Exten">
<para>Extension to use (requires <literal>Context</literal> and
<literal>Priority</literal>)</para>
</parameter>
<parameter name="Context">
<para>Context to use (requires <literal>Exten</literal> and
<literal>Priority</literal>)</para>
</parameter>
<parameter name="Priority">
<para>Priority to use (requires <literal>Exten</literal> and
<literal>Context</literal>)</para>
</parameter>
<parameter name="Application">
<para>Application to execute.</para>
</parameter>
<parameter name="Data">
<para>Data to use (requires <literal>Application</literal>).</para>
</parameter>
<parameter name="Timeout" default="30000">
<para>How long to wait for call to be answered (in ms.).</para>
</parameter>
<parameter name="CallerID">
<para>Caller ID to be set on the outgoing channel.</para>
</parameter>
<parameter name="Variable">
<para>Channel variable to set, multiple Variable: headers are allowed.</para>
</parameter>
<parameter name="Account">
<para>Account code.</para>
</parameter>
<parameter name="EarlyMedia">
<para>Set to <literal>true</literal> to force call bridge on early media..</para>
</parameter>
<parameter name="Async">
<para>Set to <literal>true</literal> for fast origination.</para>
</parameter>
<parameter name="Codecs">
<para>Comma-separated list of codecs to use for this call.</para>
</parameter>
<parameter name="ChannelId">
<para>Channel UniqueId to be set on the channel.</para>
</parameter>
<parameter name="OtherChannelId">
<para>Channel UniqueId to be set on the second local channel.</para>
</parameter>
</syntax>
<description>
<para>Generates an outgoing call to a
<replaceable>Extension</replaceable>/<replaceable>Context</replaceable>/<replaceable>Priority</replaceable>
or <replaceable>Application</replaceable>/<replaceable>Data</replaceable>
</para>
</description>
<see-also>
<ref type="managerEvent">OriginateResponse</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="OriginateResponse">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to an Originate command.</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="Response">
<enumlist>
<enum name="Failure"/>
<enum name="Success"/>
</enumlist>
</parameter>
<parameter name="Channel"/>
<parameter name="Context"/>
<parameter name="Exten"/>
<parameter name="Application"/>
<parameter name="Data"/>
<parameter name="Reason"/>
<parameter name="Uniqueid"/>
<parameter name="CallerIDNum"/>
<parameter name="CallerIDName"/>
</syntax>
<see-also>
<ref type="manager">Originate</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="Command" language="en_US">
<synopsis>
Execute Asterisk CLI Command.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Command" required="true">
<para>Asterisk CLI command to run.</para>
</parameter>
</syntax>
<description>
<para>Run a CLI command.</para>
</description>
</manager>
<manager name="ExtensionState" language="en_US">
<synopsis>
Check Extension Status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Exten" required="true">
<para>Extension to check state on.</para>
</parameter>
<parameter name="Context" required="true">
<para>Context for extension.</para>
</parameter>
</syntax>
<description>
<para>Report the extension state for given extension. If the extension has a hint,
will use devicestate to check the status of the device connected to the extension.</para>
<para>Will return an <literal>Extension Status</literal> message. The response will include
the hint for the extension and the status.</para>
</description>
<see-also>
<ref type="managerEvent">ExtensionStatus</ref>
</see-also>
</manager>
<manager name="PresenceState" language="en_US">
<synopsis>
Check Presence State
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Provider" required="true">
<para>Presence Provider to check the state of</para>
</parameter>
</syntax>
<description>
<para>Report the presence state for the given presence provider.</para>
<para>Will return a <literal>Presence State</literal> message. The response will include the
presence state and, if set, a presence subtype and custom message.</para>
</description>
<see-also>
<ref type="managerEvent">PresenceStatus</ref>
</see-also>
</manager>
<manager name="AbsoluteTimeout" language="en_US">
<synopsis>
Set absolute timeout.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to hangup.</para>
</parameter>
<parameter name="Timeout" required="true">
<para>Maximum duration of the call (sec).</para>
</parameter>
</syntax>
<description>
<para>Hangup a channel after a certain time. Acknowledges set time with
<literal>Timeout Set</literal> message.</para>
</description>
</manager>
<manager name="MailboxStatus" language="en_US">
<synopsis>
Check mailbox.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Full mailbox ID <replaceable>mailbox</replaceable>@<replaceable>vm-context</replaceable>.</para>
</parameter>
</syntax>
<description>
<para>Checks a voicemail account for status.</para>
<para>Returns whether there are messages waiting.</para>
<para>Message: Mailbox Status.</para>
<para>Mailbox: <replaceable>mailboxid</replaceable>.</para>
<para>Waiting: <literal>0</literal> if messages waiting, <literal>1</literal>
if no messages waiting.</para>
</description>
<see-also>
<ref type="manager">MailboxCount</ref>
</see-also>
</manager>
<manager name="MailboxCount" language="en_US">
<synopsis>
Check Mailbox Message Count.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Full mailbox ID <replaceable>mailbox</replaceable>@<replaceable>vm-context</replaceable>.</para>
</parameter>
</syntax>
<description>
<para>Checks a voicemail account for new messages.</para>
<para>Returns number of urgent, new and old messages.</para>
<para>Message: Mailbox Message Count</para>
<para>Mailbox: <replaceable>mailboxid</replaceable></para>
<para>UrgentMessages: <replaceable>count</replaceable></para>
<para>NewMessages: <replaceable>count</replaceable></para>
<para>OldMessages: <replaceable>count</replaceable></para>
</description>
<see-also>
<ref type="manager">MailboxStatus</ref>
</see-also>
</manager>
<manager name="ListCommands" language="en_US">
<synopsis>
List available manager commands.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Returns the action name and synopsis for every action that
is available to the user.</para>
</description>
</manager>
<manager name="SendText" language="en_US">
<synopsis>
Send text message to channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel to send message to.</para>
</parameter>
<parameter name="Message" required="true">
<para>Message to send.</para>
</parameter>
</syntax>
<description>
<para>Sends A Text Message to a channel while in a call.</para>
</description>
</manager>
<manager name="UserEvent" language="en_US">
<synopsis>
Send an arbitrary event.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="UserEvent" required="true">
<para>Event string to send.</para>
</parameter>
<parameter name="Header1">
<para>Content1.</para>
</parameter>
<parameter name="HeaderN">
<para>ContentN.</para>
</parameter>
</syntax>
<description>
<para>Send an event to manager sessions.</para>
</description>
<see-also>
<ref type="managerEvent">UserEvent</ref>
<ref type="application">UserEvent</ref>
</see-also>
</manager>
<manager name="WaitEvent" language="en_US">
<synopsis>
Wait for an event to occur.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Timeout" required="true">
<para>Maximum time (in seconds) to wait for events, <literal>-1</literal> means forever.</para>
</parameter>
</syntax>
<description>
<para>This action will ellicit a <literal>Success</literal> response. Whenever
a manager event is queued. Once WaitEvent has been called on an HTTP manager
session, events will be generated and queued.</para>
</description>
</manager>
<manager name="CoreSettings" language="en_US">
<synopsis>
Show PBX core settings (version etc).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Query for Core PBX settings.</para>
</description>
</manager>
<manager name="CoreStatus" language="en_US">
<synopsis>
Show PBX core status variables.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Query for Core PBX status.</para>
</description>
</manager>
<manager name="Reload" language="en_US">
<synopsis>
Send a reload event.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module">
<para>Name of the module to reload.</para>
</parameter>
</syntax>
<description>
<para>Send a reload event.</para>
</description>
<see-also>
<ref type="manager">ModuleLoad</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="CoreShowChannel">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a CoreShowChannels command.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<channel_snapshot/>
<parameter name="BridgeId">
<para>Identifier of the bridge the channel is in, may be empty if not in one</para>
</parameter>
<parameter name="Application">
<para>Application currently executing on the channel</para>
</parameter>
<parameter name="ApplicationData">
<para>Data given to the currently executing application</para>
</parameter>
<parameter name="Duration">
<para>The amount of time the channel has existed</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">CoreShowChannels</ref>
<ref type="managerEvent">CoreShowChannelsComplete</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="CoreShowChannelsComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised at the end of the CoreShowChannel list produced by the CoreShowChannels command.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="EventList">
<para>Conveys the status of the command reponse list</para>
</parameter>
<parameter name="ListItems">
<para>The total number of list items produced</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">CoreShowChannels</ref>
<ref type="managerEvent">CoreShowChannel</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="CoreShowChannels" language="en_US">
<synopsis>
List currently active channels.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>List currently defined channels and some information about them.</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='CoreShowChannel'])" />
</list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='CoreShowChannelsComplete'])" />
</responses>
</manager>
<manager name="LoggerRotate" language="en_US">
<synopsis>
Reload and rotate the Asterisk logger.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Reload and rotate the logger. Analogous to the CLI command 'logger rotate'.</para>
</description>
</manager>
<manager name="ModuleLoad" language="en_US">
<synopsis>
Module management.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module">
<para>Asterisk module name (including .so extension) or subsystem identifier:</para>
<enumlist>
<enum name="cdr" />
<enum name="dnsmgr" />
<enum name="extconfig" />
<enum name="enum" />
<enum name="acl" />
<enum name="manager" />
<enum name="http" />
<enum name="logger" />
<enum name="features" />
<enum name="dsp" />
<enum name="udptl" />
<enum name="indications" />
<enum name="cel" />
<enum name="plc" />
</enumlist>
</parameter>
<parameter name="LoadType" required="true">
<para>The operation to be done on module. Subsystem identifiers may only
be reloaded.</para>
<enumlist>
<enum name="load" />
<enum name="unload" />
<enum name="reload" />
</enumlist>
<para>If no module is specified for a <literal>reload</literal> loadtype,
all modules are reloaded.</para>
</parameter>
</syntax>
<description>
<para>Loads, unloads or reloads an Asterisk module in a running system.</para>
</description>
<see-also>
<ref type="manager">Reload</ref>
<ref type="manager">ModuleCheck</ref>
</see-also>
</manager>
<manager name="ModuleCheck" language="en_US">
<synopsis>
Check if module is loaded.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module" required="true">
<para>Asterisk module name (not including extension).</para>
</parameter>
</syntax>
<description>
<para>Checks if Asterisk module is loaded. Will return Success/Failure.
For success returns, the module revision number is included.</para>
</description>
<see-also>
<ref type="manager">ModuleLoad</ref>
</see-also>
</manager>
<manager name="AOCMessage" language="en_US">
<synopsis>
Generate an Advice of Charge message on a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to generate the AOC message on.</para>
</parameter>
<parameter name="ChannelPrefix">
<para>Partial channel prefix. By using this option one can match the beginning part
of a channel name without having to put the entire name in. For example
if a channel name is SIP/snom-00000001 and this value is set to SIP/snom, then
that channel matches and the message will be sent. Note however that only
the first matched channel has the message sent on it. </para>
</parameter>
<parameter name="MsgType" required="true">
<para>Defines what type of AOC message to create, AOC-D or AOC-E</para>
<enumlist>
<enum name="D" />
<enum name="E" />
</enumlist>
</parameter>
<parameter name="ChargeType" required="true">
<para>Defines what kind of charge this message represents.</para>
<enumlist>
<enum name="NA" />
<enum name="FREE" />
<enum name="Currency" />
<enum name="Unit" />
</enumlist>
</parameter>
<parameter name="UnitAmount(0)">
<para>This represents the amount of units charged. The ETSI AOC standard specifies that
this value along with the optional UnitType value are entries in a list. To accommodate this
these values take an index value starting at 0 which can be used to generate this list of
unit entries. For Example, If two unit entires were required this could be achieved by setting the
paramter UnitAmount(0)=1234 and UnitAmount(1)=5678. Note that UnitAmount at index 0 is
required when ChargeType=Unit, all other entries in the list are optional.
</para>
</parameter>
<parameter name="UnitType(0)">
<para>Defines the type of unit. ETSI AOC standard specifies this as an integer
value between 1 and 16, but this value is left open to accept any positive
integer. Like the UnitAmount parameter, this value represents a list entry
and has an index parameter that starts at 0.
</para>
</parameter>
<parameter name="CurrencyName">
<para>Specifies the currency's name. Note that this value is truncated after 10 characters.</para>
</parameter>
<parameter name="CurrencyAmount">
<para>Specifies the charge unit amount as a positive integer. This value is required
when ChargeType==Currency.</para>
</parameter>
<parameter name="CurrencyMultiplier">
<para>Specifies the currency multiplier. This value is required when ChargeType==Currency.</para>
<enumlist>
<enum name="OneThousandth" />
<enum name="OneHundredth" />
<enum name="OneTenth" />
<enum name="One" />
<enum name="Ten" />
<enum name="Hundred" />
<enum name="Thousand" />
</enumlist>
</parameter>
<parameter name="TotalType" default="Total">
<para>Defines what kind of AOC-D total is represented.</para>
<enumlist>
<enum name="Total" />
<enum name="SubTotal" />
</enumlist>
</parameter>
<parameter name="AOCBillingId">
<para>Represents a billing ID associated with an AOC-D or AOC-E message. Note
that only the first 3 items of the enum are valid AOC-D billing IDs</para>
<enumlist>
<enum name="Normal" />
<enum name="ReverseCharge" />
<enum name="CreditCard" />
<enum name="CallFwdUnconditional" />
<enum name="CallFwdBusy" />
<enum name="CallFwdNoReply" />
<enum name="CallDeflection" />
<enum name="CallTransfer" />
</enumlist>
</parameter>
<parameter name="ChargingAssociationId">
<para>Charging association identifier. This is optional for AOC-E and can be
set to any value between -32768 and 32767</para>
</parameter>
<parameter name="ChargingAssociationNumber">
<para>Represents the charging association party number. This value is optional
for AOC-E.</para>
</parameter>
<parameter name="ChargingAssociationPlan">
<para>Integer representing the charging plan associated with the ChargingAssociationNumber.
The value is bits 7 through 1 of the Q.931 octet containing the type-of-number and
numbering-plan-identification fields.</para>
</parameter>
</syntax>
<description>
<para>Generates an AOC-D or AOC-E message on a channel.</para>
</description>
<see-also>
<ref type="managerEvent">AOC-D</ref>
<ref type="managerEvent">AOC-E</ref>
</see-also>
</manager>
<function name="AMI_CLIENT" language="en_US">
<synopsis>
Checks attributes of manager accounts
</synopsis>
<syntax>
<parameter name="loginname" required="true">
<para>Login name, specified in manager.conf</para>
</parameter>
<parameter name="field" required="true">
<para>The manager account attribute to return</para>
<enumlist>
<enum name="sessions">
<para>The number of sessions for this AMI account</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>
Currently, the only supported parameter is "sessions" which will return the current number of
active sessions for this AMI account.
</para>
</description>
</function>
<manager name="Filter" language="en_US">
<synopsis>
Dynamically add filters for the current manager session.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Operation">
<enumlist>
<enum name="Add">
<para>Add a filter.</para>
</enum>
</enumlist>
</parameter>
<parameter name="Filter">
<para>Filters can be whitelist or blacklist</para>
<para>Example whitelist filter: "Event: Newchannel"</para>
<para>Example blacklist filter: "!Channel: DAHDI.*"</para>
<para>This filter option is used to whitelist or blacklist events per user to be
reported with regular expressions and are allowed if both the regex matches
and the user has read access as defined in manager.conf. Filters are assumed to be for whitelisting
unless preceeded by an exclamation point, which marks it as being black.
Evaluation of the filters is as follows:</para>
<para>- If no filters are configured all events are reported as normal.</para>
<para>- If there are white filters only: implied black all filter processed first, then white filters.</para>
<para>- If there are black filters only: implied white all filter processed first, then black filters.</para>
<para>- If there are both white and black filters: implied black all filter processed first, then white
filters, and lastly black filters.</para>
</parameter>
</syntax>
<description>
<para>The filters added are only used for the current session.
Once the connection is closed the filters are removed.</para>
<para>This comand requires the system permission because
this command can be used to create filters that may bypass
filters defined in manager.conf</para>
</description>
<see-also>
<ref type="manager">FilterList</ref>
</see-also>
</manager>
<manager name="FilterList" language="en_US">
<synopsis>
Show current event filters for this session
</synopsis>
<description>
<para>The filters displayed are for the current session. Only those filters defined in
manager.conf will be present upon starting a new session.</para>
</description>
<see-also>
<ref type="manager">Filter</ref>
</see-also>
</manager>
<manager name="BlindTransfer" language="en_US">
<synopsis>
Blind transfer channel(s) to the given destination
</synopsis>
<syntax>
<parameter name="Channel" required="true">
</parameter>
<parameter name="Context">
</parameter>
<parameter name="Exten">
</parameter>
</syntax>
<description>
<para>Redirect all channels currently bridged to the specified channel to the specified destination.</para>
</description>
<see-also>
<ref type="manager">Redirect</ref>
<ref type="managerEvent">BlindTransfer</ref>
</see-also>
</manager>
<managerEvent name="ExtensionStatus" language="en_US">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a hint changes due to a device state change.</synopsis>
<syntax>
<parameter name="Exten">
<para>Name of the extension.</para>
</parameter>
<parameter name="Context">
<para>Context that owns the extension.</para>
</parameter>
<parameter name="Hint">
<para>Hint set for the extension</para>
</parameter>
<parameter name="Status">
<para>Numerical value of the extension status. Extension
status is determined by the combined device state of all items
contained in the hint.</para>
<enumlist>
<enum name="-2">
<para>The extension was removed from the dialplan.</para>
</enum>
<enum name="-1">
<para>The extension's hint was removed from the dialplan.</para>
</enum>
<enum name="0">
<para>
<literal>Idle</literal> - Related device(s) are in an idle
state.</para>
</enum>
<enum name="1">
<para>
<literal>InUse</literal> - Related device(s) are in active
calls but may take more calls.</para>
</enum>
<enum name="2">
<para>
<literal>Busy</literal> - Related device(s) are in active
calls and may not take any more calls.</para>
</enum>
<enum name="4">
<para>
<literal>Unavailable</literal> - Related device(s) are
not reachable.</para>
</enum>
<enum name="8">
<para>
<literal>Ringing</literal> - Related device(s) are
currently ringing.</para>
</enum>
<enum name="9">
<para>
<literal>InUse&amp;Ringing</literal> - Related device(s)
are currently ringing and in active calls.</para>
</enum>
<enum name="16">
<para>
<literal>Hold</literal> - Related device(s) are
currently on hold.</para>
</enum>
<enum name="17">
<para>
<literal>InUse&amp;Hold</literal> - Related device(s)
are currently on hold and in active calls.</para>
</enum>
</enumlist>
</parameter>
<parameter name="StatusText">
<para>Text representation of <literal>Status</literal>.</para>
<enumlist>
<enum name="Idle" />
<enum name="InUse" />
<enum name="Busy" />
<enum name="Unavailable" />
<enum name="Ringing" />
<enum name="InUse&amp;Ringing" />
<enum name="Hold" />
<enum name="InUse&amp;Hold" />
<enum name="Unknown">
<para>Status does not match any of the above values.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="manager">ExtensionState</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent name="PresenceStatus" language="en_US">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a hint changes due to a presence state change.</synopsis>
<syntax>
<parameter name="Exten" />
<parameter name="Context" />
<parameter name="Hint" />
<parameter name="Status" />
<parameter name="Subtype" />
<parameter name="Message" />
</syntax>
<see-also>
<ref type="manager">PresenceState</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AOC-S">
<managerEventInstance class="EVENT_FLAG_AOC">
<synopsis>Raised when an Advice of Charge message is sent at the beginning of a call.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Chargeable" />
<parameter name="RateType">
<enumlist>
<enum name="NotAvailable" />
<enum name="Free" />
<enum name="FreeFromBeginning" />
<enum name="Duration" />
<enum name="Flag" />
<enum name="Volume" />
<enum name="SpecialCode" />
</enumlist>
</parameter>
<parameter name="Currency" />
<parameter name="Name" />
<parameter name="Cost" />
<parameter name="Multiplier">
<enumlist>
<enum name="1/1000" />
<enum name="1/100" />
<enum name="1/10" />
<enum name="1" />
<enum name="10" />
<enum name="100" />
<enum name="1000" />
</enumlist>
</parameter>
<parameter name="ChargingType" />
<parameter name="StepFunction" />
<parameter name="Granularity" />
<parameter name="Length" />
<parameter name="Scale" />
<parameter name="Unit">
<enumlist>
<enum name="Octect" />
<enum name="Segment" />
<enum name="Message" />
</enumlist>
</parameter>
<parameter name="SpecialCode" />
</syntax>
<see-also>
<ref type="managerEvent">AOC-D</ref>
<ref type="managerEvent">AOC-E</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AOC-D">
<managerEventInstance class="EVENT_FLAG_AOC">
<synopsis>Raised when an Advice of Charge message is sent during a call.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Charge" />
<parameter name="Type">
<enumlist>
<enum name="NotAvailable" />
<enum name="Free" />
<enum name="Currency" />
<enum name="Units" />
</enumlist>
</parameter>
<parameter name="BillingID">
<enumlist>
<enum name="Normal" />
<enum name="Reverse" />
<enum name="CreditCard" />
<enum name="CallForwardingUnconditional" />
<enum name="CallForwardingBusy" />
<enum name="CallForwardingNoReply" />
<enum name="CallDeflection" />
<enum name="CallTransfer" />
<enum name="NotAvailable" />
</enumlist>
</parameter>
<parameter name="TotalType">
<enumlist>
<enum name="SubTotal" />
<enum name="Total" />
</enumlist>
</parameter>
<parameter name="Currency" />
<parameter name="Name" />
<parameter name="Cost" />
<parameter name="Multiplier">
<enumlist>
<enum name="1/1000" />
<enum name="1/100" />
<enum name="1/10" />
<enum name="1" />
<enum name="10" />
<enum name="100" />
<enum name="1000" />
</enumlist>
</parameter>
<parameter name="Units" />
<parameter name="NumberOf" />
<parameter name="TypeOf" />
</syntax>
<see-also>
<ref type="manager">AOCMessage</ref>
<ref type="managerEvent">AOC-S</ref>
<ref type="managerEvent">AOC-E</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AOC-E">
<managerEventInstance class="EVENT_FLAG_AOC">
<synopsis>Raised when an Advice of Charge message is sent at the end of a call.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="ChargingAssociation" />
<parameter name="Number" />
<parameter name="Plan" />
<parameter name="ID" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AOC-D']/managerEventInstance/syntax/parameter)" />
</syntax>
<see-also>
<ref type="manager">AOCMessage</ref>
<ref type="managerEvent">AOC-S</ref>
<ref type="managerEvent">AOC-D</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<configInfo name="res_pjsip_acl" language="en_US">
<synopsis>SIP ACL module</synopsis>
<description>
<para>
<emphasis>ACL</emphasis>
</para>
<para>
The ACL module used by <literal>res_pjsip</literal>. This module is
independent of <literal>endpoints</literal> and operates on all inbound
SIP communication using res_pjsip.
</para>
<para>
There are two main ways of defining your ACL with the options
provided. You can use the <literal>permit</literal> and <literal>deny</literal> options
which act on <emphasis>IP</emphasis> addresses, or the <literal>contactpermit</literal>
and <literal>contactdeny</literal> options which act on <emphasis>Contact header</emphasis>
addresses in incoming REGISTER requests. You can combine the various options to
create a mixed ACL.
</para>
<para>
Additionally, instead of defining an ACL with options, you can reference IP or
Contact header ACLs from the file <filename>acl.conf</filename> by using the <literal>acl</literal>
or <literal>contactacl</literal> options.
</para>
</description>
<configFile name="pjsip.conf">
<configObject name="acl">
<synopsis>Access Control List</synopsis>
<configOption name="acl">
<synopsis>List of IP ACL section names in acl.conf</synopsis>
<description>
<para>
This matches sections configured in <literal>acl.conf</literal>. The value is
defined as a list of comma-delimited section names.
</para>
</description>
</configOption>
<configOption name="contact_acl">
<synopsis>List of Contact ACL section names in acl.conf</synopsis>
<description>
<para>
This matches sections configured in <literal>acl.conf</literal>. The value is
defined as a list of comma-delimited section names.
</para>
</description>
</configOption>
<configOption name="contact_deny">
<synopsis>List of Contact header addresses to deny</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="contact_permit">
<synopsis>List of Contact header addresses to permit</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="deny">
<synopsis>List of IP addresses to deny access from</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="permit">
<synopsis>List of IP addresses to permit access from</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'acl'.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_pjsip_publish_asterisk" language="en_US">
<synopsis>SIP resource for inbound and outbound Asterisk event publications</synopsis>
<description>
<para>
<emphasis>Inbound and outbound Asterisk event publication</emphasis>
</para>
<para>This module allows <literal>res_pjsip</literal> to send and receive Asterisk event publications.</para>
</description>
<configFile name="pjsip.conf">
<configObject name="asterisk-publication">
<synopsis>The configuration for inbound Asterisk event publication</synopsis>
<description>
<para>
Publish is <emphasis>COMPLETELY</emphasis> separate from the rest of
<literal>pjsip.conf</literal>.
</para>
</description>
<configOption name="devicestate_publish">
<synopsis>Optional name of a publish item that can be used to publish a request for full device state information.</synopsis>
</configOption>
<configOption name="mailboxstate_publish">
<synopsis>Optional name of a publish item that can be used to publish a request for full mailbox state information.</synopsis>
</configOption>
<configOption name="device_state" default="no">
<synopsis>Whether we should permit incoming device state events.</synopsis>
</configOption>
<configOption name="device_state_filter">
<synopsis>Optional regular expression used to filter what devices we accept events for.</synopsis>
</configOption>
<configOption name="mailbox_state" default="no">
<synopsis>Whether we should permit incoming mailbox state events.</synopsis>
</configOption>
<configOption name="mailbox_state_filter">
<synopsis>Optional regular expression used to filter what mailboxes we accept events for.</synopsis>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'asterisk-publication'.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_pjsip" language="en_US">
<synopsis>SIP Resource using PJProject</synopsis>
<configFile name="pjsip.conf">
<configObject name="endpoint">
<synopsis>Endpoint</synopsis>
<description>
<para>
The <emphasis>Endpoint</emphasis> is the primary configuration object.
It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
dialable entries of their own. Communication with another SIP device is
accomplished via Addresses of Record (AoRs) which have one or more
contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
use a <literal>transport</literal> will default to first transport found
in <filename>pjsip.conf</filename> that matches its type.
</para>
<para>Example: An Endpoint has been configured with no transport.
When it comes time to call an AoR, PJSIP will find the
first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
will use the first IPv6 transport and try to send the request.
</para>
<para>If the anonymous endpoint identifier is in use an endpoint with the name
"anonymous@domain" will be searched for as a last resort. If this is not found
it will fall back to searching for "anonymous". If neither endpoints are found
the anonymous endpoint identifier will not return an endpoint and anonymous
calling will not be possible.
</para>
</description>
<configOption name="100rel" default="yes">
<synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
<description>
<enumlist>
<enum name="no" />
<enum name="required" />
<enum name="yes" />
</enumlist>
</description>
</configOption>
<configOption name="aggregate_mwi" default="yes">
<synopsis>Condense MWI notifications into a single NOTIFY.</synopsis>
<description>
<para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
individual NOTIFYs are sent for each mailbox.</para>
</description>
</configOption>
<configOption name="allow">
<synopsis>Media Codec(s) to allow</synopsis>
</configOption>
<configOption name="allow_overlap" default="yes">
<synopsis>Enable RFC3578 overlap dialing support.</synopsis>
</configOption>
<configOption name="aors">
<synopsis>AoR(s) to be used with the endpoint</synopsis>
<description>
<para>
List of comma separated AoRs that the endpoint should be associated with.
</para>
</description>
</configOption>
<configOption name="auth">
<synopsis>Authentication Object(s) associated with the endpoint</synopsis>
<description>
<para>
This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
</para>
<para>
Endpoints without an authentication object
configured will allow connections without verification.</para>
<note>
<para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses. See the auth realm description for details.
</para>
</note>
</description>
</configOption>
<configOption name="callerid">
<synopsis>CallerID information for the endpoint</synopsis>
<description>
<para>
Must be in the format <literal>Name &lt;Number&gt;</literal>,
or only <literal>&lt;Number&gt;</literal>.
</para>
</description>
</configOption>
<configOption name="callerid_privacy">
<synopsis>Default privacy level</synopsis>
<description>
<enumlist>
<enum name="allowed_not_screened" />
<enum name="allowed_passed_screen" />
<enum name="allowed_failed_screen" />
<enum name="allowed" />
<enum name="prohib_not_screened" />
<enum name="prohib_passed_screen" />
<enum name="prohib_failed_screen" />
<enum name="prohib" />
<enum name="unavailable" />
</enumlist>
</description>
</configOption>
<configOption name="callerid_tag">
<synopsis>Internal id_tag for the endpoint</synopsis>
</configOption>
<configOption name="context">
<synopsis>Dialplan context for inbound sessions</synopsis>
</configOption>
<configOption name="direct_media_glare_mitigation" default="none">
<synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
<description>
<para>
This setting attempts to avoid creating INVITE glare scenarios
by disabling direct media reINVITEs in one direction thereby allowing
designated servers (according to this option) to initiate direct
media reINVITEs without contention and significantly reducing call
setup time.
</para>
<para>
A more detailed description of how this option functions can be found on
the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
</para>
<enumlist>
<enum name="none" />
<enum name="outgoing" />
<enum name="incoming" />
</enumlist>
</description>
</configOption>
<configOption name="direct_media_method" default="invite">
<synopsis>Direct Media method type</synopsis>
<description>
<para>Method for setting up Direct Media between endpoints.</para>
<enumlist>
<enum name="invite" />
<enum name="reinvite">
<para>Alias for the <literal>invite</literal> value.</para>
</enum>
<enum name="update" />
</enumlist>
</description>
</configOption>
<configOption name="connected_line_method" default="invite">
<synopsis>Connected line method type</synopsis>
<description>
<para>Method used when updating connected line information.</para>
<enumlist>
<enum name="invite">
<para>When set to <literal>invite</literal>, check the remote's Allow header and
if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP
renegotiation. If UPDATE is not Allowed, send INVITE.</para>
</enum>
<enum name="reinvite">
<para>Alias for the <literal>invite</literal> value.</para>
</enum>
<enum name="update">
<para>If set to <literal>update</literal>, send UPDATE regardless of what the remote
Allows. </para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="direct_media" default="yes">
<synopsis>Determines whether media may flow directly between endpoints.</synopsis>
</configOption>
<configOption name="disable_direct_media_on_nat" default="no">
<synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
</configOption>
<configOption name="disallow">
<synopsis>Media Codec(s) to disallow</synopsis>
</configOption>
<configOption name="dtmf_mode" default="rfc4733">
<synopsis>DTMF mode</synopsis>
<description>
<para>This setting allows to choose the DTMF mode for endpoint communication.</para>
<enumlist>
<enum name="rfc4733">
<para>DTMF is sent out of band of the main audio stream. This
supercedes the older <emphasis>RFC-2833</emphasis> used within
the older <literal>chan_sip</literal>.</para>
</enum>
<enum name="inband">
<para>DTMF is sent as part of audio stream.</para>
</enum>
<enum name="info">
<para>DTMF is sent as SIP INFO packets.</para>
</enum>
<enum name="auto">
<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
</enum>
<enum name="auto_info">
<para>DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="media_address">
<synopsis>IP address used in SDP for media handling</synopsis>
<description>
<para>
At the time of SDP creation, the IP address defined here will be used as
the media address for individual streams in the SDP.
</para>
<note>
<para>
Be aware that the <literal>external_media_address</literal> option, set in Transport
configuration, can also affect the final media address used in the SDP.
</para>
</note>
</description>
</configOption>
<configOption name="bind_rtp_to_media_address">
<synopsis>Bind the RTP instance to the media_address</synopsis>
<description>
<para>
If media_address is specified, this option causes the RTP instance to be bound to the
specified ip address which causes the packets to be sent from that address.
</para>
</description>
</configOption>
<configOption name="force_rport" default="yes">
<synopsis>Force use of return port</synopsis>
</configOption>
<configOption name="ice_support" default="no">
<synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
</configOption>
<configOption name="identify_by" default="username,location">
<synopsis>Way(s) for Endpoint to be identified</synopsis>
<description>
<para>
Endpoints and aors can be identified in multiple ways. Currently, the supported
options are <literal>username</literal>, which matches the endpoint or aor id based on
the username and domain in the From header (or To header for aors), and
<literal>auth_username</literal>, which matches the endpoint or aor id based on the
username and realm in the Authentication header. In all cases, if an exact match
on both username and domain/realm fails, the match will be retried with just the username.
</para>
<note>
<para>
Identification by auth_username has some security considerations because an
Authentication header is not present on the first message of a dialog when
digest authentication is used. The client can't generate it until the server
sends the challenge in a 401 response. Since Asterisk normally sends a security
event when an incoming request can't be matched to an endpoint, using auth_username
requires that the security event be deferred until a request is received with
the Authentication header and only generated if the username doesn't result in a
match. This may result in a delay before an attack is recognized. You can control
how many unmatched requests are received from a single ip address before a security
event is generated using the unidentified_request parameters in the "global"
configuration object.
</para>
</note>
<note>
<para>Endpoints can also be identified by IP address; however, that method
of identification is not handled by this configuration option. See the documentation
for the <literal>identify</literal> configuration section for more details on that
method of endpoint identification. If this option is set and an <literal>identify</literal>
configuration section exists for the endpoint, then the endpoint can be identified in
multiple ways.</para>
</note>
<enumlist>
<enum name="username" />
<enum name="auth_username" />
</enumlist>
</description>
</configOption>
<configOption name="redirect_method">
<synopsis>How redirects received from an endpoint are handled</synopsis>
<description>
<para>
When a redirect is received from an endpoint there are multiple ways it can be handled.
If this option is set to <literal>user</literal> the user portion of the redirect target
is treated as an extension within the dialplan and dialed using a Local channel. If this option
is set to <literal>uri_core</literal> the target URI is returned to the dialing application
which dials it using the PJSIP channel driver and endpoint originally used. If this option is
set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
and also supporting multiple potential redirect targets. The con is that since redirection occurs
within chan_pjsip redirecting information is not forwarded and redirection can not be
prevented.
</para>
<enumlist>
<enum name="user" />
<enum name="uri_core" />
<enum name="uri_pjsip" />
</enumlist>
</description>
</configOption>
<configOption name="mailboxes">
<synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
<description>
<para>
Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
changes happen for any of the specified mailboxes. More than one mailbox can be
specified with a comma-delimited string. app_voicemail mailboxes must be specified
as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
external sources, such as through the res_external_mwi module, you must specify
strings supported by the external system.
</para>
<para>
For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
configuration.
</para>
</description>
</configOption>
<configOption name="mwi_subscribe_replaces_unsolicited">
<synopsis>An MWI subscribe will replace sending unsolicited NOTIFYs</synopsis>
</configOption>
<configOption name="voicemail_extension">
<synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
</configOption>
<configOption name="moh_suggest" default="default">
<synopsis>Default Music On Hold class</synopsis>
</configOption>
<configOption name="outbound_auth">
<synopsis>Authentication object(s) used for outbound requests</synopsis>
<description>
<para>
This is a comma-delimited list of <replaceable>auth</replaceable>
sections defined in <filename>pjsip.conf</filename> used to respond
to outbound connection authentication challenges.</para>
<note>
<para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses. See the auth realm description for details.
</para>
</note>
</description>
</configOption>
<configOption name="outbound_proxy">
<synopsis>Full SIP URI of the outbound proxy used to send requests</synopsis>
</configOption>
<configOption name="rewrite_contact">
<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
<description>
<para>
On inbound SIP messages from this endpoint, the Contact header or an
appropriate Record-Route header will be changed to have the source IP
address and port. This option does not affect outbound messages sent to
this endpoint. This option helps servers communicate with endpoints
that are behind NATs. This option also helps reuse reliable transport
connections such as TCP and TLS.
</para>
</description>
</configOption>
<configOption name="rtp_ipv6" default="no">
<synopsis>Allow use of IPv6 for RTP traffic</synopsis>
</configOption>
<configOption name="rtp_symmetric" default="no">
<synopsis>Enforce that RTP must be symmetric</synopsis>
</configOption>
<configOption name="send_diversion" default="yes">
<synopsis>Send the Diversion header, conveying the diversion
information to the called user agent</synopsis>
</configOption>
<configOption name="send_pai" default="no">
<synopsis>Send the P-Asserted-Identity header</synopsis>
</configOption>
<configOption name="send_rpid" default="no">
<synopsis>Send the Remote-Party-ID header</synopsis>
</configOption>
<configOption name="rpid_immediate" default="no">
<synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
<description>
<para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
or <emphasis>183 Progress</emphasis> response messages to the
caller if the connected line information is updated before
the call is answered. This can send a <emphasis>180 Ringing</emphasis>
response before the call has even reached the far end. The
caller can start hearing ringback before the far end even gets
the call. Many phones tend to grab the first connected line
information and refuse to update the display if it changes. The
first information is not likely to be correct if the call
goes to an endpoint not under the control of this Asterisk
box.</para>
<para>When disabled, a connected line update must wait for
another reason to send a message with the connected line
information to the caller before the call is answered. You can
trigger the sending of the information by using an appropriate
dialplan application such as <emphasis>Ringing</emphasis>.</para>
</description>
</configOption>
<configOption name="timers_min_se" default="90">
<synopsis>Minimum session timers expiration period</synopsis>
<description>
<para>
Minimium session timer expiration period. Time in seconds.
</para>
</description>
</configOption>
<configOption name="timers" default="yes">
<synopsis>Session timers for SIP packets</synopsis>
<description>
<enumlist>
<enum name="no" />
<enum name="yes" />
<enum name="required" />
<enum name="always" />
<enum name="forced">
<para>Alias of always</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="timers_sess_expires" default="1800">
<synopsis>Maximum session timer expiration period</synopsis>
<description>
<para>
Maximium session timer expiration period. Time in seconds.
</para>
</description>
</configOption>
<configOption name="transport">
<synopsis>Desired transport configuration</synopsis>
<description>
<para>
This will set the desired transport configuration to send SIP data through.
</para>
<warning>
<para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
to the first configured transport in <filename>pjsip.conf</filename> which is
valid for the URI we are trying to contact.
</para>
</warning>
<warning>
<para>Transport configuration is not affected by reloads. In order to
change transports, a full Asterisk restart is required</para>
</warning>
</description>
</configOption>
<configOption name="trust_id_inbound" default="no">
<synopsis>Accept identification information received from this endpoint</synopsis>
<description>
<para>This option determines whether Asterisk will accept
identification from the endpoint from headers such as P-Asserted-Identity
or Remote-Party-ID header. This option applies both to calls originating from the
endpoint and calls originating from Asterisk. If <literal>no</literal>, the
configured Caller-ID from pjsip.conf will always be used as the identity for
the endpoint.</para>
</description>
</configOption>
<configOption name="trust_id_outbound" default="no">
<synopsis>Send private identification details to the endpoint.</synopsis>
<description>
<para>This option determines whether res_pjsip will send private
identification information to the endpoint. If <literal>no</literal>,
private Caller-ID information will not be forwarded to the endpoint.
"Private" in this case refers to any method of restricting identification.
Example: setting <replaceable>callerid_privacy</replaceable> to any
<literal>prohib</literal> variation.
Example: If <replaceable>trust_id_inbound</replaceable> is set to
<literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
header in a SIP request or response would indicate the identification
provided in the request is private.</para>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'endpoint'.</synopsis>
</configOption>
<configOption name="use_ptime" default="no">
<synopsis>Use Endpoint's requested packetisation interval</synopsis>
</configOption>
<configOption name="use_avpf" default="no">
<synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
endpoint.</synopsis>
<description>
<para>
If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
profile for all media offers on outbound calls and media updates and will
decline media offers not using the AVPF or SAVPF profile.
</para>
<para>
If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
profile for all media offers on outbound calls and media updates, and will
decline media offers not using the AVP or SAVP profile.
</para>
</description>
</configOption>
<configOption name="force_avp" default="no">
<synopsis>Determines whether res_pjsip will use and enforce usage of AVP,
regardless of the RTP profile in use for this endpoint.</synopsis>
<description>
<para>
If set to <literal>yes</literal>, res_pjsip will use the AVP, AVPF, SAVP, or
SAVPF RTP profile for all media offers on outbound calls and media updates including
those for DTLS-SRTP streams.
</para>
<para>
If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
depending on configuration.
</para>
</description>
</configOption>
<configOption name="media_use_received_transport" default="no">
<synopsis>Determines whether res_pjsip will use the media transport received in the
offer SDP in the corresponding answer SDP.</synopsis>
<description>
<para>
If set to <literal>yes</literal>, res_pjsip will use the received media transport.
</para>
<para>
If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
depending on configuration.
</para>
</description>
</configOption>
<configOption name="media_encryption" default="no">
<synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
for this endpoint.</synopsis>
<description>
<enumlist>
<enum name="no">
<para>
res_pjsip will offer no encryption and allow no encryption to be setup.
</para>
</enum>
<enum name="sdes">
<para>
res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
transport should be used in conjunction with this option to prevent
exposure of media encryption keys.
</para>
</enum>
<enum name="dtls">
<para>
res_pjsip will offer DTLS-SRTP setup.
</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="media_encryption_optimistic" default="no">
<synopsis>Determines whether encryption should be used if possible but does not terminate the
session if not achieved.</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>sdes</literal> or <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="g726_non_standard" default="no">
<synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
<description>
<para>
When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
packing order instead of what is recommended by RFC3551. Since this essentially
replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
specified in the endpoint's allowed codec list.
</para>
</description>
</configOption>
<configOption name="inband_progress" default="no">
<synopsis>Determines whether chan_pjsip will indicate ringing using inband
progress.</synopsis>
<description>
<para>
If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
when told to indicate ringing and will immediately start sending ringing
as audio.
</para>
<para>
If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
to indicate ringing and will NOT send it as audio.
</para>
</description>
</configOption>
<configOption name="call_group">
<synopsis>The numeric pickup groups for a channel.</synopsis>
<description>
<para>
Can be set to a comma separated list of numbers or ranges between the values
of 0-63 (maximum of 64 groups).
</para>
</description>
</configOption>
<configOption name="pickup_group">
<synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
<description>
<para>
Can be set to a comma separated list of numbers or ranges between the values
of 0-63 (maximum of 64 groups).
</para>
</description>
</configOption>
<configOption name="named_call_group">
<synopsis>The named pickup groups for a channel.</synopsis>
<description>
<para>
Can be set to a comma separated list of case sensitive strings limited by
supported line length.
</para>
</description>
</configOption>
<configOption name="named_pickup_group">
<synopsis>The named pickup groups that a channel can pickup.</synopsis>
<description>
<para>
Can be set to a comma separated list of case sensitive strings limited by
supported line length.
</para>
</description>
</configOption>
<configOption name="device_state_busy_at" default="0">
<synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
<description>
<para>
When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
PJSIP channel driver will return busy as the device state instead of in use.
</para>
</description>
</configOption>
<configOption name="t38_udptl" default="no">
<synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
<description>
<para>
If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
and relayed.
</para>
</description>
</configOption>
<configOption name="t38_udptl_ec" default="none">
<synopsis>T.38 UDPTL error correction method</synopsis>
<description>
<enumlist>
<enum name="none">
<para>
No error correction should be used.
</para>
</enum>
<enum name="fec">
<para>
Forward error correction should be used.
</para>
</enum>
<enum name="redundancy">
<para>
Redundacy error correction should be used.
</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="t38_udptl_maxdatagram" default="0">
<synopsis>T.38 UDPTL maximum datagram size</synopsis>
<description>
<para>
This option can be set to override the maximum datagram of a remote endpoint for broken
endpoints.
</para>
</description>
</configOption>
<configOption name="fax_detect" default="no">
<synopsis>Whether CNG tone detection is enabled</synopsis>
<description>
<para>
This option can be set to send the session to the fax extension when a CNG tone is
detected.
</para>
</description>
</configOption>
<configOption name="fax_detect_timeout">
<synopsis>How long into a call before fax_detect is disabled for the call</synopsis>
<description>
<para>
The option determines how many seconds into a call before the
fax_detect option is disabled for the call. Setting the value
to zero disables the timeout.
</para>
</description>
</configOption>
<configOption name="t38_udptl_nat" default="no">
<synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
<description>
<para>
When enabled the UDPTL stack will send UDPTL packets to the source address of
received packets.
</para>
</description>
</configOption>
<configOption name="t38_udptl_ipv6" default="no">
<synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
<description>
<para>
When enabled the UDPTL stack will use IPv6.
</para>
</description>
</configOption>
<configOption name="tone_zone">
<synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
</configOption>
<configOption name="language">
<synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
</configOption>
<configOption name="one_touch_recording" default="no">
<synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
<see-also>
<ref type="configOption">record_on_feature</ref>
<ref type="configOption">record_off_feature</ref>
</see-also>
</configOption>
<configOption name="record_on_feature" default="automixmon">
<synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
<description>
<para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
feature will be enabled for the channel. The feature designated here can be any built-in
or dynamic feature defined in features.conf.</para>
<note>
<para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para>
</note>
</description>
<see-also>
<ref type="configOption">one_touch_recording</ref>
<ref type="configOption">record_off_feature</ref>
</see-also>
</configOption>
<configOption name="record_off_feature" default="automixmon">
<synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
<description>
<para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
feature will be enabled for the channel. The feature designated here can be any built-in
or dynamic feature defined in features.conf.</para>
<note>
<para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para>
</note>
</description>
<see-also>
<ref type="configOption">one_touch_recording</ref>
<ref type="configOption">record_on_feature</ref>
</see-also>
</configOption>
<configOption name="rtp_engine" default="asterisk">
<synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
</configOption>
<configOption name="allow_transfer" default="yes">
<synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
</configOption>
<configOption name="user_eq_phone" default="no">
<synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
</configOption>
<configOption name="sdp_owner" default="-">
<synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
</configOption>
<configOption name="sdp_session" default="Asterisk">
<synopsis>String used for the SDP session (s=) line.</synopsis>
</configOption>
<configOption name="tos_audio">
<synopsis>DSCP TOS bits for audio streams</synopsis>
<description>
<para>
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
</para>
</description>
</configOption>
<configOption name="tos_video">
<synopsis>DSCP TOS bits for video streams</synopsis>
<description>
<para>
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
</para>
</description>
</configOption>
<configOption name="cos_audio">
<synopsis>Priority for audio streams</synopsis>
<description>
<para>
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
</para>
</description>
</configOption>
<configOption name="cos_video">
<synopsis>Priority for video streams</synopsis>
<description>
<para>
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
</para>
</description>
</configOption>
<configOption name="allow_subscribe" default="yes">
<synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
</configOption>
<configOption name="sub_min_expiry" default="60">
<synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
</configOption>
<configOption name="from_user">
<synopsis>Username to use in From header for requests to this endpoint.</synopsis>
</configOption>
<configOption name="mwi_from_user">
<synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
</configOption>
<configOption name="from_domain">
<synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
</configOption>
<configOption name="dtls_verify">
<synopsis>Verify that the provided peer certificate is valid</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="dtls_rekey">
<synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
<para>
If this is not set or the value provided is 0 rekeying will be disabled.
</para>
</description>
</configOption>
<configOption name="dtls_cert_file">
<synopsis>Path to certificate file to present to peer</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="dtls_private_key">
<synopsis>Path to private key for certificate file</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="dtls_cipher">
<synopsis>Cipher to use for DTLS negotiation</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
<para>Many options for acceptable ciphers. See link for more:</para>
<para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
</para>
</description>
</configOption>
<configOption name="dtls_ca_file">
<synopsis>Path to certificate authority certificate</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="dtls_ca_path">
<synopsis>Path to a directory containing certificate authority certificates</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="dtls_setup">
<synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
<enumlist>
<enum name="active">
<para>
res_pjsip will make a connection to the peer.
</para>
</enum>
<enum name="passive">
<para>
res_pjsip will accept connections from the peer.
</para>
</enum>
<enum name="actpass">
<para>
res_pjsip will offer and accept connections from the peer.
</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="dtls_fingerprint">
<synopsis>Type of hash to use for the DTLS fingerprint in the SDP.</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>dtls</literal>.
</para>
<enumlist>
<enum name="SHA-256"></enum>
<enum name="SHA-1"></enum>
</enumlist>
</description>
</configOption>
<configOption name="srtp_tag_32">
<synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
<description>
<para>
This option only applies if <replaceable>media_encryption</replaceable> is
set to <literal>sdes</literal> or <literal>dtls</literal>.
</para>
</description>
</configOption>
<configOption name="set_var">
<synopsis>Variable set on a channel involving the endpoint.</synopsis>
<description>
<para>
When a new channel is created using the endpoint set the specified
variable(s) on that channel. For multiple channel variables specify
multiple 'set_var'(s).
</para>
</description>
</configOption>
<configOption name="message_context">
<synopsis>Context to route incoming MESSAGE requests to.</synopsis>
<description>
<para>
If specified, incoming MESSAGE requests will be routed to the indicated
dialplan context. If no <replaceable>message_context</replaceable> is
specified, then the <replaceable>context</replaceable> setting is used.
</para>
</description>
</configOption>
<configOption name="accountcode">
<synopsis>An accountcode to set automatically on any channels created for this endpoint.</synopsis>
<description>
<para>
If specified, any channel created for this endpoint will automatically
have this accountcode set on it.
</para>
</description>
</configOption>
<configOption name="rtp_keepalive">
<synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
<description>
<para>
At the specified interval, Asterisk will send an RTP comfort noise frame. This may
be useful for situations where Asterisk is behind a NAT or firewall and must keep a
hole open in order to allow for media to arrive at Asterisk.
</para>
</description>
</configOption>
<configOption name="rtp_timeout" default="0">
<synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
<description>
<para>
This option configures the number of seconds without RTP (while off hold) before
considering a channel as dead. When the number of seconds is reached the underlying
channel is hung up. By default this option is set to 0, which means do not check.
</para>
</description>
</configOption>
<configOption name="rtp_timeout_hold" default="0">
<synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
<description>
<para>
This option configures the number of seconds without RTP (while on hold) before
considering a channel as dead. When the number of seconds is reached the underlying
channel is hung up. By default this option is set to 0, which means do not check.
</para>
</description>
</configOption>
<configOption name="acl">
<synopsis>List of IP ACL section names in acl.conf</synopsis>
<description>
<para>
This matches sections configured in <literal>acl.conf</literal>. The value is
defined as a list of comma-delimited section names.
</para>
</description>
</configOption>
<configOption name="deny">
<synopsis>List of IP addresses to deny access from</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="permit">
<synopsis>List of IP addresses to permit access from</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="contact_acl">
<synopsis>List of Contact ACL section names in acl.conf</synopsis>
<description>
<para>
This matches sections configured in <literal>acl.conf</literal>. The value is
defined as a list of comma-delimited section names.
</para>
</description>
</configOption>
<configOption name="contact_deny">
<synopsis>List of Contact header addresses to deny</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="contact_permit">
<synopsis>List of Contact header addresses to permit</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dotted-decimal notation. Separate the IP address and subnet
mask with a slash ('/')
</para>
</description>
</configOption>
<configOption name="subscribe_context">
<synopsis>Context for incoming MESSAGE requests.</synopsis>
<description>
<para>
If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context.
If no <replaceable>subscribe_context</replaceable> is specified,
then the <replaceable>context</replaceable> setting is used.
</para>
</description>
</configOption>
<configOption name="contact_user" default="">
<synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
<description>
<para>
On outbound requests, force the user portion of the Contact header to this value.
</para>
</description>
</configOption>
<configOption name="asymmetric_rtp_codec" default="no">
<synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
<description>
<para>
When set to "yes" the codec in use for sending will be allowed to differ from
that of the received one. PJSIP will not automatically switch the sending one
to the receiving one.
</para>
</description>
</configOption>
<configOption name="rtcp_mux" default="no">
<synopsis>Enable RFC 5761 RTCP multiplexing on the RTP port</synopsis>
<description>
<para>
With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux"
attribute on all media streams. This will result in RTP and RTCP being sent and received
on the same port. This shifts the demultiplexing logic to the application rather than
the transport layer. This option is useful when interoperating with WebRTC endpoints
since they mandate this option's use.
</para>
</description>
</configOption>
<configOption name="refer_blind_progress" default="yes">
<synopsis>Whether to notifies all the progress details on blind transfer</synopsis>
<description>
<para>
Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK"
after REFER has been accepted. If set to <literal>no</literal> then asterisk
will not send the progress details, but immediately will send "200 OK".
</para>
</description>
</configOption>
<configOption name="notify_early_inuse_ringing" default="no">
<synopsis>Whether to notifies dialog-info 'early' on InUse&amp;Ringing state</synopsis>
<description>
<para>
Control whether dialog-info subscriptions get 'early' state
on Ringing when already INUSE.
</para>
</description>
</configOption>
<configOption name="incoming_mwi_mailbox">
<synopsis>Mailbox name to use when incoming MWI NOTIFYs are received</synopsis>
<description>
<para>
If an MWI NOTIFY is received <emphasis>from</emphasis> this endpoint,
this mailbox will be used when notifying other modules of MWI status
changes. If not set, incoming MWI NOTIFYs are ignored.
</para>
</description>
</configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>
<description>
<para>
Authentication objects hold the authentication information for use
by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
This also allows for multiple objects to use a single auth object. See
the <literal>auth_type</literal> config option for password style choices.
</para>
</description>
<configOption name="auth_type" default="userpass">
<synopsis>Authentication type</synopsis>
<description>
<para>
This option specifies which of the password style config options should be read
when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
then we'll read from the 'password' option. For <literal>md5</literal> we'll read
from 'md5_cred'.
</para>
<enumlist>
<enum name="md5"/>
<enum name="userpass"/>
</enumlist>
</description>
</configOption>
<configOption name="nonce_lifetime" default="32">
<synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
</configOption>
<configOption name="md5_cred">
<synopsis>MD5 Hash used for authentication.</synopsis>
<description>
<para>Only used when auth_type is <literal>md5</literal>.</para>
</description>
</configOption>
<configOption name="password">
<synopsis>PlainText password used for authentication.</synopsis>
<description>
<para>Only used when auth_type is <literal>userpass</literal>.</para>
</description>
</configOption>
<configOption name="realm">
<synopsis>SIP realm for endpoint</synopsis>
<description>
<para>
The treatment of this value depends upon how the authentication
object is used.
</para>
<para>
When used as an inbound authentication object, the realm is sent
as part of the challenge so the peer can know which key to use
when responding. An empty value will use the
<replaceable>global</replaceable> section's
<literal>default_realm</literal> value when issuing a challenge.
</para>
<para>
When used as an outbound authentication object, the realm is
matched with the received challenge realm to determine which
authentication object to use when responding to the challenge. An
empty value matches any challenging realm when determining
which authentication object matches a received challenge.
</para>
<note>
<para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses.</para>
</note>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be 'auth'</synopsis>
</configOption>
<configOption name="username">
<synopsis>Username to use for account</synopsis>
</configOption>
</configObject>
<configObject name="domain_alias">
<synopsis>Domain Alias</synopsis>
<description>
<para>
Signifies that a domain is an alias. If the domain on a session is
not found to match an AoR then this object is used to see if we have
an alias for the AoR to which the endpoint is binding. This objects
name as defined in configuration should be the domain alias and a
config option is provided to specify the domain to be aliased.
</para>
</description>
<configOption name="type">
<synopsis>Must be of type 'domain_alias'.</synopsis>
</configOption>
<configOption name="domain">
<synopsis>Domain to be aliased</synopsis>
</configOption>
</configObject>
<configObject name="transport">
<synopsis>SIP Transport</synopsis>
<description>
<para>
<emphasis>Transports</emphasis>
</para>
<para>There are different transports and protocol derivatives
supported by <literal>res_pjsip</literal>. They are in order of
preference: UDP, TCP, and WebSocket (WS).</para>
<note>
<para>Changes to transport configuration in pjsip.conf will only be
effected on a complete restart of Asterisk. A module reload
will not suffice.</para>
</note>
</description>
<configOption name="async_operations" default="1">
<synopsis>Number of simultaneous Asynchronous Operations</synopsis>
</configOption>
<configOption name="bind">
<synopsis>IP Address and optional port to bind to for this transport</synopsis>
</configOption>
<configOption name="ca_list_file">
<synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
</configOption>
<configOption name="ca_list_path">
<synopsis>Path to directory containing a list of certificates to read (TLS ONLY)</synopsis>
</configOption>
<configOption name="cert_file">
<synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
<description>
<para>
A path to a .crt or .pem file can be provided. However, only
the certificate is read from the file, not the private key.
The <literal>priv_key_file</literal> option must supply a
matching key file.
</para>
</description>
</configOption>
<configOption name="cipher">
<synopsis>Preferred cryptography cipher names (TLS ONLY)</synopsis>
<description>
<para>Comma separated list of cipher names or numeric equivalents.
Numeric equivalents can be either decimal or hexadecimal (0xX).
</para>
<para>There are many cipher names. Use the CLI command
<literal>pjsip list ciphers</literal> to see a list of cipher
names available for your installation. See link for more:</para>
<para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES
</para>
</description>
</configOption>
<configOption name="domain">
<synopsis>Domain the transport comes from</synopsis>
</configOption>
<configOption name="external_media_address">
<synopsis>External IP address to use in RTP handling</synopsis>
<description>
<para>
When a request or response is sent out, if the destination of the
message is outside the IP network defined in the option <literal>localnet</literal>,
and the media address in the SDP is within the localnet network, then the
media address in the SDP will be rewritten to the value defined for
<literal>external_media_address</literal>.
</para>
</description>
</configOption>
<configOption name="external_signaling_address">
<synopsis>External address for SIP signalling</synopsis>
</configOption>
<configOption name="external_signaling_port" default="0">
<synopsis>External port for SIP signalling</synopsis>
</configOption>
<configOption name="method">
<synopsis>Method of SSL transport (TLS ONLY)</synopsis>
<description>
<enumlist>
<enum name="default">
<para>The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.</para>
</enum>
<enum name="unspecified">
<para>This option is equivalent to setting 'default'</para>
</enum>
<enum name="tlsv1" />
<enum name="sslv2" />
<enum name="sslv3" />
<enum name="sslv23" />
</enumlist>
</description>
</configOption>
<configOption name="local_net">
<synopsis>Network to consider local (used for NAT purposes).</synopsis>
<description>
<para>This must be in CIDR or dotted decimal format with the IP
and mask separated with a slash ('/').</para>
</description>
</configOption>
<configOption name="password">
<synopsis>Password required for transport</synopsis>
</configOption>
<configOption name="priv_key_file">
<synopsis>Private key file (TLS ONLY)</synopsis>
</configOption>
<configOption name="protocol" default="udp">
<synopsis>Protocol to use for SIP traffic</synopsis>
<description>
<enumlist>
<enum name="udp" />
<enum name="tcp" />
<enum name="tls" />
<enum name="ws" />
<enum name="wss" />
</enumlist>
</description>
</configOption>
<configOption name="require_client_cert" default="false">
<synopsis>Require client certificate (TLS ONLY)</synopsis>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'transport'.</synopsis>
</configOption>
<configOption name="verify_client" default="false">
<synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
</configOption>
<configOption name="verify_server" default="false">
<synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
</configOption>
<configOption name="tos" default="false">
<synopsis>Enable TOS for the signalling sent over this transport</synopsis>
<description>
<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
for more information on this parameter.</para>
<note>
<para>This option does not apply to the <replaceable>ws</replaceable>
or the <replaceable>wss</replaceable> protocols.</para>
</note>
</description>
</configOption>
<configOption name="cos" default="false">
<synopsis>Enable COS for the signalling sent over this transport</synopsis>
<description>
<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
for more information on this parameter.</para>
<note>
<para>This option does not apply to the <replaceable>ws</replaceable>
or the <replaceable>wss</replaceable> protocols.</para>
</note>
</description>
</configOption>
<configOption name="websocket_write_timeout">
<synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
<description>
<para>If a websocket connection accepts input slowly, the timeout
for writes to it can be increased to keep it from being disconnected.
Value is in milliseconds; default is 100 ms.</para>
</description>
</configOption>
<configOption name="allow_reload" default="no">
<synopsis>Allow this transport to be reloaded.</synopsis>
<description>
<para>Allow this transport to be reloaded when res_pjsip is reloaded.
This option defaults to "no" because reloading a transport may disrupt
in-progress calls.</para>
</description>
</configOption>
<configOption name="symmetric_transport" default="no">
<synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis>
<description>
<para>When a request from a dynamic contact
comes in on a transport with this option set to 'yes',
the transport name will be saved and used for subsequent
outgoing requests like OPTIONS, NOTIFY and INVITE. It's
saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output.
On the outgoing request, if a transport wasn't explicitly
set on the endpoint AND the request URI is not a hostname,
the saved transport will be used and the 'x-ast-txp'
parameter stripped from the outgoing packet.
</para>
</description>
</configOption>
</configObject>
<configObject name="contact">
<synopsis>A way of creating an aliased name to a SIP URI</synopsis>
<description>
<para>
Contacts are a way to hide SIP URIs from the dialplan directly.
They are also used to make a group of contactable parties when
in use with <literal>AoR</literal> lists.
</para>
</description>
<configOption name="type">
<synopsis>Must be of type 'contact'.</synopsis>
</configOption>
<configOption name="uri">
<synopsis>SIP URI to contact peer</synopsis>
</configOption>
<configOption name="expiration_time">
<synopsis>Time to keep alive a contact</synopsis>
<description>
<para>
Time to keep alive a contact. String style specification.
</para>
</description>
</configOption>
<configOption name="qualify_frequency" default="0">
<synopsis>Interval at which to qualify a contact</synopsis>
<description>
<para>
Interval between attempts to qualify the contact for reachability.
If <literal>0</literal> never qualify. Time in seconds.
</para>
</description>
</configOption>
<configOption name="qualify_timeout" default="3.0">
<synopsis>Timeout for qualify</synopsis>
<description>
<para>
If the contact doesn't repond to the OPTIONS request before the timeout,
the contact is marked unavailable.
If <literal>0</literal> no timeout. Time in fractional seconds.
</para>
</description>
</configOption>
<configOption name="authenticate_qualify" default="no">
<synopsis>Authenticates a qualify request if needed</synopsis>
<description>
<para>
If true and a qualify request receives a challenge or authenticate response
authentication is attempted before declaring the contact available.
</para>
</description>
</configOption>
<configOption name="outbound_proxy">
<synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
<description>
<para>
If set the provided URI will be used as the outbound proxy when an
OPTIONS request is sent to a contact for qualify purposes.
</para>
</description>
</configOption>
<configOption name="path">
<synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
</configOption>
<configOption name="user_agent">
<synopsis>User-Agent header from registration.</synopsis>
<description>
<para>
The User-Agent is automatically stored based on data present in incoming SIP
REGISTER requests and is not intended to be configured manually.
</para>
</description>
</configOption>
<configOption name="endpoint">
<synopsis>Endpoint name</synopsis>
<description>
<para>
The name of the endpoint this contact belongs to
</para>
</description>
</configOption>
<configOption name="reg_server">
<synopsis>Asterisk Server name</synopsis>
<description>
<para>
Asterisk Server name on which SIP endpoint registered.
</para>
</description>
</configOption>
<configOption name="via_addr">
<synopsis>IP-address of the last Via header from registration.</synopsis>
<description>
<para>
The last Via header should contain the address of UA which sent the request.
The IP-address of the last Via header is automatically stored based on data present
in incoming SIP REGISTER requests and is not intended to be configured manually.
</para>
</description>
</configOption>
<configOption name="via_port">
<synopsis>IP-port of the last Via header from registration.</synopsis>
<description>
<para>
The IP-port of the last Via header is automatically stored based on data present
in incoming SIP REGISTER requests and is not intended to be configured manually.
</para>
</description>
</configOption>
<configOption name="call_id">
<synopsis>Call-ID header from registration.</synopsis>
<description>
<para>
The Call-ID header is automatically stored based on data present
in incoming SIP REGISTER requests and is not intended to be configured manually.
</para>
</description>
</configOption>
<configOption name="prune_on_boot">
<synopsis>A contact that cannot survive a restart/boot.</synopsis>
<description>
<para>
The option is set if the incoming SIP REGISTER contact is rewritten
on a reliable transport and is not intended to be configured manually.
</para>
</description>
</configOption>
</configObject>
<configObject name="aor">
<synopsis>The configuration for a location of an endpoint</synopsis>
<description>
<para>
An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
AoRs are specified, an endpoint will not be reachable by Asterisk.
Beyond that, an AoR has other uses within Asterisk, such as inbound
registration.
</para>
<para>
An <literal>AoR</literal> is a way to allow dialing a group
of <literal>Contacts</literal> that all use the same
<literal>endpoint</literal> for calls.
</para>
<para>
This can be used as another way of grouping a list of contacts to dial
rather than specifing them each directly when dialing via the dialplan.
This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
</para>
<para>
Registrations: For Asterisk to match an inbound registration to an endpoint,
the AoR object name must match the user portion of the SIP URI in the "To:"
header of the inbound SIP registration. That will usually be equivalent
to the "user name" set in your hard or soft phones configuration.
</para>
</description>
<configOption name="contact">
<synopsis>Permanent contacts assigned to AoR</synopsis>
<description>
<para>
Contacts specified will be called whenever referenced
by <literal>chan_pjsip</literal>.
</para>
<para>
Use a separate "contact=" entry for each contact required. Contacts
are specified using a SIP URI.
</para>
</description>
</configOption>
<configOption name="default_expiration" default="3600">
<synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
</configOption>
<configOption name="mailboxes">
<synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
<description>
<para>This option applies when an external entity subscribes to an AoR
for Message Waiting Indications. The mailboxes specified will be subscribed to.
More than one mailbox can be specified with a comma-delimited string.
app_voicemail mailboxes must be specified as mailbox@context;
for example: mailboxes=6001@default. For mailboxes provided by external sources,
such as through the res_external_mwi module, you must specify strings supported by
the external system.
</para>
<para>
For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
</para>
</description>
</configOption>
<configOption name="voicemail_extension">
<synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
</configOption>
<configOption name="maximum_expiration" default="7200">
<synopsis>Maximum time to keep an AoR</synopsis>
<description>
<para>
Maximium time to keep a peer with explicit expiration. Time in seconds.
</para>
</description>
</configOption>
<configOption name="max_contacts" default="0">
<synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
<description>
<para>
Maximum number of contacts that can associate with this AoR. This value does
not affect the number of contacts that can be added with the "contact" option.
It only limits contacts added through external interaction, such as
registration.
</para>
<note>
<para>The <replaceable>rewrite_contact</replaceable> option
registers the source address as the contact address to help with
NAT and reusing connection oriented transports such as TCP and
TLS. Unfortunately, refreshing a registration may register a
different contact address and exceed
<replaceable>max_contacts</replaceable>. The
<replaceable>remove_existing</replaceable> option can help by
removing the soonest to expire contact(s) over
<replaceable>max_contacts</replaceable> which is likely the
old <replaceable>rewrite_contact</replaceable> contact source
address being refreshed.
</para>
</note>
<note>
<para>This should be set to <literal>1</literal> and
<replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
wish to stick with the older <literal>chan_sip</literal> behaviour.
</para>
</note>
</description>
</configOption>
<configOption name="minimum_expiration" default="60">
<synopsis>Minimum keep alive time for an AoR</synopsis>
<description>
<para>
Minimum time to keep a peer with an explicit expiration. Time in seconds.
</para>
</description>
</configOption>
<configOption name="remove_existing" default="no">
<synopsis>Determines whether new contacts replace existing ones.</synopsis>
<description>
<para>
On receiving a new registration to the AoR should it remove enough
existing contacts not added or updated by the registration to
satisfy <replaceable>max_contacts</replaceable>? Any removed
contacts will expire the soonest.
</para>
<note>
<para>The <replaceable>rewrite_contact</replaceable> option
registers the source address as the contact address to help with
NAT and reusing connection oriented transports such as TCP and
TLS. Unfortunately, refreshing a registration may register a
different contact address and exceed
<replaceable>max_contacts</replaceable>. The
<replaceable>remove_existing</replaceable> option can help by
removing the soonest to expire contact(s) over
<replaceable>max_contacts</replaceable> which is likely the
old <replaceable>rewrite_contact</replaceable> contact source
address being refreshed.
</para>
</note>
<note>
<para>This should be set to <literal>yes</literal> and
<replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
wish to stick with the older <literal>chan_sip</literal> behaviour.
</para>
</note>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'aor'.</synopsis>
</configOption>
<configOption name="qualify_frequency" default="0">
<synopsis>Interval at which to qualify an AoR</synopsis>
<description>
<para>
Interval between attempts to qualify the AoR for reachability.
If <literal>0</literal> never qualify. Time in seconds.
</para>
</description>
</configOption>
<configOption name="qualify_timeout" default="3.0">
<synopsis>Timeout for qualify</synopsis>
<description>
<para>
If the contact doesn't repond to the OPTIONS request before the timeout,
the contact is marked unavailable.
If <literal>0</literal> no timeout. Time in fractional seconds.
</para>
</description>
</configOption>
<configOption name="authenticate_qualify" default="no">
<synopsis>Authenticates a qualify request if needed</synopsis>
<description>
<para>
If true and a qualify request receives a challenge or authenticate response
authentication is attempted before declaring the contact available.
</para>
</description>
</configOption>
<configOption name="outbound_proxy">
<synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
<description>
<para>
If set the provided URI will be used as the outbound proxy when an
OPTIONS request is sent to a contact for qualify purposes.
</para>
</description>
</configOption>
<configOption name="support_path">
<synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
<description>
<para>
When this option is enabled, the Path headers in register requests will be saved
and its contents will be used in Route headers for outbound out-of-dialog requests
and in Path headers for outbound 200 responses. Path support will also be indicated
in the Supported header.
</para>
</description>
</configOption>
</configObject>
<configObject name="system">
<synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
<description>
<para>
The settings in this section are global. In addition to being global, the values will
not be re-evaluated when a reload is performed. This is because the values must be set
before the SIP stack is initialized. The only way to reset these values is to either
restart Asterisk, or unload res_pjsip.so and then load it again.
</para>
</description>
<configOption name="timer_t1" default="500">
<synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
<description>
<para>
Timer T1 is the base for determining how long to wait before retransmitting
requests that receive no response when using an unreliable transport (e.g. UDP).
For more information on this timer, see RFC 3261, Section 17.1.1.1.
</para>
</description>
</configOption>
<configOption name="timer_b" default="32000">
<synopsis>Set transaction timer B value (milliseconds).</synopsis>
<description>
<para>
Timer B determines the maximum amount of time to wait after sending an INVITE
request before terminating the transaction. It is recommended that this be set
to 64 * Timer T1, but it may be set higher if desired. For more information on
this timer, see RFC 3261, Section 17.1.1.1.
</para>
</description>
</configOption>
<configOption name="compact_headers" default="no">
<synopsis>Use the short forms of common SIP header names.</synopsis>
</configOption>
<configOption name="threadpool_initial_size" default="0">
<synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
</configOption>
<configOption name="threadpool_auto_increment" default="5">
<synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
</configOption>
<configOption name="threadpool_idle_timeout" default="60">
<synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
</configOption>
<configOption name="threadpool_max_size" default="0">
<synopsis>Maximum number of threads in the res_pjsip threadpool.
A value of 0 indicates no maximum.</synopsis>
</configOption>
<configOption name="disable_tcp_switch" default="yes">
<synopsis>Disable automatic switching from UDP to TCP transports.</synopsis>
<description>
<para>
Disable automatic switching from UDP to TCP transports if outgoing
request is too large. See RFC 3261 section 18.1.1.
</para>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'system'.</synopsis>
</configOption>
</configObject>
<configObject name="global">
<synopsis>Options that apply globally to all SIP communications</synopsis>
<description>
<para>
The settings in this section are global. Unlike options in the <literal>system</literal>
section, these options can be refreshed by performing a reload.
</para>
</description>
<configOption name="max_forwards" default="70">
<synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
</configOption>
<configOption name="keep_alive_interval" default="0">
<synopsis>The interval (in seconds) to send keepalives to active connection-oriented transports.</synopsis>
</configOption>
<configOption name="contact_expiration_check_interval" default="30">
<synopsis>The interval (in seconds) to check for expired contacts.</synopsis>
</configOption>
<configOption name="disable_multi_domain" default="no">
<synopsis>Disable Multi Domain support</synopsis>
<description>
<para>
If disabled it can improve realtime performace by reducing number of database requsts.
</para>
</description>
</configOption>
<configOption name="max_initial_qualify_time" default="0">
<synopsis>The maximum amount of time from startup that qualifies should be attempted on all contacts.
If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.</synopsis>
</configOption>
<configOption name="unidentified_request_period" default="5">
<synopsis>The number of seconds over which to accumulate unidentified requests.</synopsis>
<description>
<para>
If <literal>unidentified_request_count</literal> unidentified requests are received
during <literal>unidentified_request_period</literal>, a security event will be generated.
</para>
</description>
</configOption>
<configOption name="unidentified_request_count" default="5">
<synopsis>The number of unidentified requests from a single IP to allow.</synopsis>
<description>
<para>
If <literal>unidentified_request_count</literal> unidentified requests are received
during <literal>unidentified_request_period</literal>, a security event will be generated.
</para>
</description>
</configOption>
<configOption name="unidentified_request_prune_interval" default="30">
<synopsis>The interval at which unidentified requests are older than
twice the unidentified_request_period are pruned.</synopsis>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'global'.</synopsis>
</configOption>
<configOption name="user_agent" default="Asterisk &lt;Asterisk Version&gt;">
<synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
</configOption>
<configOption name="regcontext" default="">
<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
peer who registers or unregisters with us.</synopsis>
</configOption>
<configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
<synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
</configOption>
<configOption name="default_voicemail_extension">
<synopsis>The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor</synopsis>
</configOption>
<configOption name="debug" default="no">
<synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
a host address</synopsis>
</configOption>
<configOption name="endpoint_identifier_order" default="ip,username,anonymous">
<synopsis>The order by which endpoint identifiers are processed and checked.
Identifier names are usually derived from and can be found in the endpoint
identifier module itself (res_pjsip_endpoint_identifier_*).
You can use the CLI command "pjsip show identifiers" to see the
identifiers currently available.</synopsis>
<description>
<note>
<para>
One of the identifiers is "auth_username" which matches on the username in
an Authentication header. This method has some security considerations because an
Authentication header is not present on the first message of a dialog when
digest authentication is used. The client can't generate it until the server
sends the challenge in a 401 response. Since Asterisk normally sends a security
event when an incoming request can't be matched to an endpoint, using auth_username
requires that the security event be deferred until a request is received with
the Authentication header and only generated if the username doesn't result in a
match. This may result in a delay before an attack is recognized. You can control
how many unmatched requests are received from a single ip address before a security
event is generated using the unidentified_request parameters.
</para>
</note>
</description>
</configOption>
<configOption name="default_from_user" default="asterisk">
<synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
set to this value if there is no better option (such as CallerID) to be
used.</synopsis>
</configOption>
<configOption name="default_realm" default="asterisk">
<synopsis>When Asterisk generates a challenge, the digest realm will be
set to this value if there is no better option (such as auth/realm) to be
used.</synopsis>
</configOption>
<configOption name="mwi_tps_queue_high" default="500">
<synopsis>MWI taskprocessor high water alert trigger level.</synopsis>
<description>
<para>On a heavily loaded system you may need to adjust the
taskprocessor queue limits. If any taskprocessor queue size
reaches its high water level then pjsip will stop processing
new requests until the alert is cleared. The alert clears
when all alerting taskprocessor queues have dropped to their
low water clear level.
</para>
</description>
</configOption>
<configOption name="mwi_tps_queue_low" default="-1">
<synopsis>MWI taskprocessor low water clear alert level.</synopsis>
<description>
<para>On a heavily loaded system you may need to adjust the
taskprocessor queue limits. If any taskprocessor queue size
reaches its high water level then pjsip will stop processing
new requests until the alert is cleared. The alert clears
when all alerting taskprocessor queues have dropped to their
low water clear level.
</para>
<note>
<para>Set to -1 for the low water level to be 90% of
the high water level.</para>
</note>
</description>
</configOption>
<configOption name="mwi_disable_initial_unsolicited" default="no">
<synopsis>Enable/Disable sending unsolicited MWI to all endpoints on startup.</synopsis>
<description>
<para>When the initial unsolicited MWI notification are
enabled on startup then the initial notifications
get sent at startup. If you have a lot of endpoints
(thousands) that use unsolicited MWI then you may
want to consider disabling the initial startup
notifications.
</para>
<para>When the initial unsolicited MWI notifications are
disabled on startup then the notifications will start
on the endpoint's next contact update.
</para>
</description>
</configOption>
<configOption name="ignore_uri_user_options">
<synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
<description>
<para>If you have this option enabled and there are semicolons
in the user field of a SIP URI then the field is truncated
at the first semicolon. This effectively makes the semicolon
a non-usable character for PJSIP endpoint names, extensions,
and AORs. This can be useful for improving compatability with
an ITSP that likes to use user options for whatever reason.
</para>
<example title="Sample SIP URI">
sip:1235557890;phone-context=national@x.x.x.x;user=phone
</example>
<example title="Sample SIP URI user field">
1235557890;phone-context=national
</example>
<example title="Sample SIP URI user field truncated">
1235557890
</example>
<note>
<para>The caller-id and redirecting number strings
obtained from incoming SIP URI user fields are always truncated
at the first semicolon.</para>
</note>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="PJSIPQualify" language="en_US">
<synopsis>
Qualify a chan_pjsip endpoint.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Endpoint" required="true">
<para>The endpoint you want to qualify.</para>
</parameter>
</syntax>
<description>
<para>Qualify a chan_pjsip endpoint.</para>
</description>
</manager>
<managerEvent language="en_US" name="IdentifyDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about an identify section.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'identify'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="Endpoint">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='endpoint']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Match">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match']/synopsis/node())"/>
</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the endpoint associated with this information.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AorDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'aor'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="MinimumExpiration">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MaximumExpiration">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DefaultExpiration">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/>
</para>
</parameter>
<parameter name="QualifyFrequency">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AuthenticateQualify">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MaxContacts">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RemoveExisting">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Mailboxes">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/>
</para>
</parameter>
<parameter name="OutboundProxy">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SupportPath">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TotalContacts">
<para>The total number of contacts associated with this AoR.</para>
</parameter>
<parameter name="ContactsRegistered">
<para>The number of non-permanent contacts associated with this AoR.</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the endpoint associated with this information.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AuthDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about an authentication section.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'auth'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="Username">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Password">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Md5Cred">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Realm">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/>
</para>
</parameter>
<parameter name="NonceLifetime">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AuthType">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/>
</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the endpoint associated with this information.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="TransportDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about an authentication section.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'transport'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="Protocol">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='protocol']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Bind">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='bind']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AsycOperations">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='async_operations']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CaListFile">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_file']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CaListPath">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_path']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CertFile">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cert_file']/synopsis/node())"/>
</para>
</parameter>
<parameter name="PrivKeyFile">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='priv_key_file']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Password">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='password']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ExternalSignalingAddress">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_address']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ExternalSignalingPort">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_port']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ExternalMediaAddress">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_media_address']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Domain">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='domain']/synopsis/node())"/>
</para>
</parameter>
<parameter name="VerifyServer">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_server']/synopsis/node())"/>
</para>
</parameter>
<parameter name="VerifyClient">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_client']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RequireClientCert">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='require_client_cert']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Method">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='method']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Cipher">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cipher']/synopsis/node())"/>
</para>
</parameter>
<parameter name="LocalNet">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='local_net']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Tos">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='tos']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Cos">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cos']/synopsis/node())"/>
</para>
</parameter>
<parameter name="WebsocketWriteTimeout">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='websocket_write_timeout']/synopsis/node())"/>
</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the endpoint associated with this information.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="EndpointDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about an endpoint section.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'endpoint'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="Context">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='context']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Disallow">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disallow']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Allow">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtmfMode">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtmf_mode']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RtpIpv6">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_ipv6']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RtpSymmetric">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_symmetric']/synopsis/node())"/>
</para>
</parameter>
<parameter name="IceSupport">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='ice_support']/synopsis/node())"/>
</para>
</parameter>
<parameter name="UsePtime">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_ptime']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ForceRport">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_rport']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RewriteContact">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rewrite_contact']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Transport">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='transport']/synopsis/node())"/>
</para>
</parameter>
<parameter name="OutboundProxy">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_proxy']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MohSuggest">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_suggest']/synopsis/node())"/>
</para>
</parameter>
<parameter name="100rel">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='100rel']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Timers">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TimersMinSe">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_min_se']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TimersSessExpires">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_sess_expires']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Auth">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='auth']/synopsis/node())"/>
</para>
</parameter>
<parameter name="OutboundAuth">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_auth']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Aors">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aors']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MediaAddress">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_address']/synopsis/node())"/>
</para>
</parameter>
<parameter name="IdentifyBy">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='identify_by']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DirectMedia">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DirectMediaMethod">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_method']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ConnectedLineMethod">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='connected_line_method']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DirectMediaGlareMitigation">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_glare_mitigation']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DisableDirectMediaOnNat">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disable_direct_media_on_nat']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Callerid">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CalleridPrivacy">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_privacy']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CalleridTag">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_tag']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TrustIdInbound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_inbound']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TrustIdOutbound">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_outbound']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SendPai">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_pai']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SendRpid">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_rpid']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SendDiversion">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_diversion']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Mailboxes">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mailboxes']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AggregateMwi">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aggregate_mwi']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MediaEncryption">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MediaEncryptionOptimistic">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption_optimistic']/synopsis/node())"/>
</para>
</parameter>
<parameter name="UseAvpf">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_avpf']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ForceAvp">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_avp']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MediaUseReceivedTransport">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_use_received_transport']/synopsis/node())"/>
</para>
</parameter>
<parameter name="OneTouchRecording">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='one_touch_recording']/synopsis/node())"/>
</para>
</parameter>
<parameter name="InbandProgress">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='inband_progress']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CallGroup">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='call_group']/synopsis/node())"/>
</para>
</parameter>
<parameter name="PickupGroup">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='pickup_group']/synopsis/node())"/>
</para>
</parameter>
<parameter name="NamedCallGroup">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_call_group']/synopsis/node())"/>
</para>
</parameter>
<parameter name="NamedPickupGroup">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_pickup_group']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DeviceStateBusyAt">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='device_state_busy_at']/synopsis/node())"/>
</para>
</parameter>
<parameter name="T38Udptl">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl']/synopsis/node())"/>
</para>
</parameter>
<parameter name="T38UdptlEc">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ec']/synopsis/node())"/>
</para>
</parameter>
<parameter name="T38UdptlMaxdatagram">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_maxdatagram']/synopsis/node())"/>
</para>
</parameter>
<parameter name="FaxDetect">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='fax_detect']/synopsis/node())"/>
</para>
</parameter>
<parameter name="T38UdptlNat">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_nat']/synopsis/node())"/>
</para>
</parameter>
<parameter name="T38UdptlIpv6">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ipv6']/synopsis/node())"/>
</para>
</parameter>
<parameter name="ToneZone">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tone_zone']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Language">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='language']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RecordOnFeature">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_on_feature']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RecordOffFeature">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_off_feature']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AllowTransfer">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/>
</para>
</parameter>
<parameter name="UserEqPhone">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SdpOwner">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SdpSession">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_session']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TosAudio">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_audio']/synopsis/node())"/>
</para>
</parameter>
<parameter name="TosVideo">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_video']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CosAudio">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_audio']/synopsis/node())"/>
</para>
</parameter>
<parameter name="CosVideo">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_video']/synopsis/node())"/>
</para>
</parameter>
<parameter name="AllowSubscribe">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_subscribe']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SubMinExpiry">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sub_min_expiry']/synopsis/node())"/>
</para>
</parameter>
<parameter name="FromUser">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_user']/synopsis/node())"/>
</para>
</parameter>
<parameter name="FromDomain">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_domain']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MwiFromUser">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mwi_from_user']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RtpEngine">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_engine']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsVerify">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_verify']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsRekey">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_rekey']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsCertFile">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cert_file']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsPrivateKey">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_private_key']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsCipher">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cipher']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsCaFile">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_file']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsCaPath">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_path']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DtlsSetup">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_setup']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SrtpTag32">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='srtp_tag_32']/synopsis/node())"/>
</para>
</parameter>
<parameter name="RedirectMethod">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='redirect_method']/synopsis/node())"/>
</para>
</parameter>
<parameter name="SetVar">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='set_var']/synopsis/node())"/>
</para>
</parameter>
<parameter name="MessageContext">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='message_context']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Accountcode">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/>
</para>
</parameter>
<parameter name="DeviceState">
<para>The aggregate device state for this endpoint.</para>
</parameter>
<parameter name="ActiveChannels">
<para>The number of active channels associated with this endpoint.</para>
</parameter>
<parameter name="SubscribeContext">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='subscribe_context']/synopsis/node())"/>
</para>
</parameter>
<parameter name="Allowoverlap">
<para>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_overlap']/synopsis/node())"/>
</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ContactStatusDetail">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about a contact's status.</synopsis>
<syntax>
<parameter name="AOR">
<para>The AoR that owns this contact.</para>
</parameter>
<parameter name="URI">
<para>This contact's URI.</para>
</parameter>
<parameter name="Status">
<para>This contact's status.</para>
<enumlist>
<enum name="Reachable"/>
<enum name="Unreachable"/>
</enumlist>
</parameter>
<parameter name="RoundtripUsec">
<para>The round trip time in microseconds.</para>
</parameter>
<parameter name="EndpointName">
<para>The name of the endpoint associated with this information.</para>
</parameter>
<parameter name="UserAgent">
<para>Content of the User-Agent header in REGISTER request</para>
</parameter>
<parameter name="RegExpire">
<para>Absolute time that this contact is no longer valid after</para>
</parameter>
<parameter name="ViaAddress">
<para>IP address:port of the last Via header in REGISTER request.
Will only appear in the event if available.</para>
</parameter>
<parameter name="CallID">
<para>Content of the Call-ID header in REGISTER request.
Will only appear in the event if available.</para>
</parameter>
<parameter name="ID">
<para>The sorcery ID of the contact.</para>
</parameter>
<parameter name="AuthenticateQualify">
<para>A boolean indicating whether a qualify should be authenticated.</para>
</parameter>
<parameter name="OutboundProxy">
<para>The contact's outbound proxy.</para>
</parameter>
<parameter name="Path">
<para>The Path header received on the REGISTER.</para>
</parameter>
<parameter name="QualifyFrequency">
<para>The interval in seconds at which the contact will be qualified.</para>
</parameter>
<parameter name="QualifyTimeout">
<para>The elapsed time in decimal seconds after which an OPTIONS
message is sent before the contact is considered unavailable.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="EndpointList">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide details about a contact's status.</synopsis>
<syntax>
<parameter name="ObjectType">
<para>The object's type. This will always be 'endpoint'.</para>
</parameter>
<parameter name="ObjectName">
<para>The name of this object.</para>
</parameter>
<parameter name="Transport">
<para>The transport configurations associated with this endpoint.</para>
</parameter>
<parameter name="Aor">
<para>The aor configurations associated with this endpoint.</para>
</parameter>
<parameter name="Auths">
<para>The inbound authentication configurations associated with this endpoint.</para>
</parameter>
<parameter name="OutboundAuths">
<para>The outbound authentication configurations associated with this endpoint.</para>
</parameter>
<parameter name="DeviceState">
<para>The aggregate device state for this endpoint.</para>
</parameter>
<parameter name="ActiveChannels">
<para>The number of active channels associated with this endpoint.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="PJSIPShowEndpoints" language="en_US">
<synopsis>
Lists PJSIP endpoints.
</synopsis>
<syntax />
<description>
<para>
Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
is raised that contains relevant attributes and status information. Once all
endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointList'])" />
</list-elements>
<managerEvent language="en_US" name="EndpointListComplete">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide final information about an endpoint list.</synopsis>
<syntax>
<parameter name="EventList"/>
<parameter name="ListItems"/>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<manager name="PJSIPShowEndpoint" language="en_US">
<synopsis>
Detail listing of an endpoint and its objects.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Endpoint" required="true">
<para>The endpoint to list.</para>
</parameter>
</syntax>
<description>
<para>
Provides a detailed listing of options for a given endpoint. Events are issued
showing the configuration and status of the endpoint and associated objects. These
events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
<literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
<literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
associated (for instance AoRs). Once all detail events have been raised a final
<literal>EndpointDetailComplete</literal> event is issued.
</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointDetail'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='IdentifyDetail'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactStatusDetail'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthDetail'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='TransportDetail'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AorDetail'])" />
</list-elements>
<managerEvent language="en_US" name="EndpointDetailComplete">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>Provide final information about endpoint details.</synopsis>
<syntax>
<parameter name="EventList"/>
<parameter name="ListItems"/>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<function name="ODBC" language="en_US">
<synopsis>
Controls ODBC transaction properties.
</synopsis>
<syntax>
<parameter name="property" required="true">
<enumlist>
<enum name="transaction">
<para>Gets or sets the active transaction ID. If set, and the transaction ID does not
exist and a <replaceable>database name</replaceable> is specified as an argument, it will be created.</para>
</enum>
<enum name="forcecommit">
<para>Controls whether a transaction will be automatically committed when the channel
hangs up. Defaults to false. If a <replaceable>transaction ID</replaceable> is specified in the optional argument,
the property will be applied to that ID, otherwise to the current active ID.</para>
</enum>
<enum name="isolation">
<para>Controls the data isolation on uncommitted transactions. May be one of the
following: <literal>read_committed</literal>, <literal>read_uncommitted</literal>,
<literal>repeatable_read</literal>, or <literal>serializable</literal>. Defaults to the
database setting in <filename>res_odbc.conf</filename> or <literal>read_committed</literal>
if not specified. If a <replaceable>transaction ID</replaceable> is specified as an optional argument, it will be
applied to that ID, otherwise the current active ID.</para>
</enum>
</enumlist>
</parameter>
<parameter name="argument" required="false" />
</syntax>
<description>
<para>The ODBC() function allows setting several properties to influence how a connected
database processes transactions.</para>
</description>
</function>
<application name="ODBC_Commit" language="en_US">
<synopsis>
Commits a currently open database transaction.
</synopsis>
<syntax>
<parameter name="transaction ID" required="no" />
</syntax>
<description>
<para>Commits the database transaction specified by <replaceable>transaction ID</replaceable>
or the current active transaction, if not specified.</para>
</description>
</application>
<application name="ODBC_Rollback" language="en_US">
<synopsis>
Rollback a currently open database transaction.
</synopsis>
<syntax>
<parameter name="transaction ID" required="no" />
</syntax>
<description>
<para>Rolls back the database transaction specified by <replaceable>transaction ID</replaceable>
or the current active transaction, if not specified.</para>
</description>
</application>
<configInfo name="res_pjsip_outbound_registration" language="en_US">
<synopsis>SIP resource for outbound registrations</synopsis>
<description>
<para>
<emphasis>Outbound Registration</emphasis>
</para>
<para>This module allows <literal>res_pjsip</literal> to register to other SIP servers.</para>
</description>
<configFile name="pjsip.conf">
<configObject name="registration">
<synopsis>The configuration for outbound registration</synopsis>
<description>
<para>
Registration is <emphasis>COMPLETELY</emphasis> separate from the rest of
<literal>pjsip.conf</literal>. A minimal configuration consists of
setting a <literal>server_uri</literal> and a <literal>client_uri</literal>.
</para>
</description>
<configOption name="auth_rejection_permanent" default="yes">
<synopsis>Determines whether failed authentication challenges are treated
as permanent failures.</synopsis>
<description>
<para>If this option is enabled and an authentication challenge fails,
registration will not be attempted again until the configuration is reloaded.</para>
</description>
</configOption>
<configOption name="client_uri">
<synopsis>Client SIP URI used when attemping outbound registration</synopsis>
<description>
<para>
This is the address-of-record for the outbound registration (i.e. the URI in
the To header of the REGISTER).</para>
<para>For registration with an ITSP, the client SIP URI may need to consist of
an account name or number and the provider's hostname for their registrar, e.g.
client_uri=1234567890@example.com. This may differ between providers.</para>
<para>For registration to generic registrars, the client SIP URI will depend
on networking specifics and configuration of the registrar.
</para>
</description>
</configOption>
<configOption name="contact_user">
<synopsis>Contact User to use in request</synopsis>
</configOption>
<configOption name="expiration" default="3600">
<synopsis>Expiration time for registrations in seconds</synopsis>
</configOption>
<configOption name="max_retries" default="10">
<synopsis>Maximum number of registration attempts.</synopsis>
</configOption>
<configOption name="outbound_auth" default="">
<synopsis>Authentication object(s) to be used for outbound registrations.</synopsis>
<description>
<para>
This is a comma-delimited list of <replaceable>auth</replaceable>
sections defined in <filename>pjsip.conf</filename> used to respond
to outbound authentication challenges.</para>
<note>
<para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses. See the auth realm description for details.
</para>
</note>
</description>
</configOption>
<configOption name="outbound_proxy" default="">
<synopsis>Full SIP URI of the outbound proxy used to send registrations</synopsis>
</configOption>
<configOption name="retry_interval" default="60">
<synopsis>Interval in seconds between retries if outbound registration is unsuccessful</synopsis>
</configOption>
<configOption name="forbidden_retry_interval" default="0">
<synopsis>Interval used when receiving a 403 Forbidden response.</synopsis>
<description>
<para>
If a 403 Forbidden is received, chan_pjsip will wait
<replaceable>forbidden_retry_interval</replaceable> seconds before
attempting registration again. If 0 is specified, chan_pjsip will not
retry after receiving a 403 Forbidden response. Setting this to a non-zero
value goes against a "SHOULD NOT" in RFC3261, but can be used to work around
buggy registrars.
</para>
</description>
</configOption>
<configOption name="fatal_retry_interval" default="0">
<synopsis>Interval used when receiving a Fatal response.</synopsis>
<description>
<para>
If a fatal response is received, chan_pjsip will wait
<replaceable>fatal_retry_interval</replaceable> seconds before
attempting registration again. If 0 is specified, chan_pjsip will not
retry after receiving a fatal (non-temporary 4xx, 5xx, 6xx) response.
Setting this to a non-zero value may go against a "SHOULD NOT" in RFC3261,
but can be used to work around buggy registrars.</para>
<note>
<para>if also set the <replaceable>forbidden_retry_interval</replaceable>
takes precedence over this one when a 403 is received.
Also, if <replaceable>auth_rejection_permanent</replaceable> equals 'yes' then
a 401 and 407 become subject to this retry interval.</para>
</note>
</description>
</configOption>
<configOption name="server_uri">
<synopsis>SIP URI of the server to register against</synopsis>
<description>
<para>
This is the URI at which to find the registrar to send the outbound REGISTER. This URI
is used as the request URI of the outbound REGISTER request from Asterisk.</para>
<para>For registration with an ITSP, the setting may often be just the domain of
the registrar, e.g. sip:sip.example.com.
</para>
</description>
</configOption>
<configOption name="transport">
<synopsis>Transport used for outbound authentication</synopsis>
<description>
<note>
<para>A <replaceable>transport</replaceable> configured in
<literal>pjsip.conf</literal>. As with other <literal>res_pjsip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para>
</note>
</description>
</configOption>
<configOption name="line">
<synopsis>Whether to add a 'line' parameter to the Contact for inbound call matching</synopsis>
<description>
<para>
When enabled this option will cause a 'line' parameter to be added to the Contact
header placed into the outgoing registration request. If the remote server sends a call
this line parameter will be used to establish a relationship to the outbound registration,
ultimately causing the configured endpoint to be used.
</para>
</description>
</configOption>
<configOption name="endpoint">
<synopsis>Endpoint to use for incoming related calls</synopsis>
<description>
<para>
When line support is enabled this configured endpoint name is used for incoming calls
that are related to the outbound registration.
</para>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'registration'.</synopsis>
</configOption>
<configOption name="support_path">
<synopsis>Enables Path support for outbound REGISTER requests.</synopsis>
<description>
<para>
When this option is enabled, outbound REGISTER requests will advertise
support for Path headers so that intervening proxies can add to the Path
header as necessary.
</para>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="PJSIPUnregister" language="en_US">
<synopsis>
Unregister an outbound registration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Registration" required="true">
<para>The outbound registration to unregister or '*all' to unregister them all.</para>
</parameter>
</syntax>
<description>
<para>
Unregisters the specified (or all) outbound registration(s) and stops future registration attempts.
Call PJSIPRegister to start registration and schedule re-registrations according to configuration.
</para>
</description>
</manager>
<manager name="PJSIPRegister" language="en_US">
<synopsis>
Register an outbound registration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Registration" required="true">
<para>The outbound registration to register or '*all' to register them all.</para>
</parameter>
</syntax>
<description>
<para>
Unregisters the specified (or all) outbound registration(s) then starts registration and schedules re-registrations
according to configuration.
</para>
</description>
</manager>
<manager name="PJSIPShowRegistrationsOutbound" language="en_US">
<synopsis>
Lists PJSIP outbound registrations.
</synopsis>
<syntax />
<description>
<para>
In response <literal>OutboundRegistrationDetail</literal> events showing configuration and status
information are raised for each outbound registration object. <literal>AuthDetail</literal>
events are raised for each associated auth object as well. Once all events are completed an
<literal>OutboundRegistrationDetailComplete</literal> is issued.
</para>
</description>
</manager>
<configInfo name="res_hep" language="en_US">
<synopsis>Resource for integration with Homer using HEPv3</synopsis>
<configFile name="hep.conf">
<configObject name="general">
<synopsis>General settings.</synopsis>
<description>
<para>
The <emphasis>general</emphasis> settings section contains information
to configure Asterisk as a Homer capture agent.
</para>
</description>
<configOption name="enabled" default="yes">
<synopsis>Enable or disable packet capturing.</synopsis>
<description>
<enumlist>
<enum name="no" />
<enum name="yes" />
</enumlist>
</description>
</configOption>
<configOption name="uuid_type" default="call-id">
<synopsis>The preferred type of UUID to pass to Homer.</synopsis>
<description>
<enumlist>
<enum name="call-id">
<para>Use the PJSIP Call-Id</para>
</enum>
<enum name="channel">
<para>Use the Asterisk channel name</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="capture_address">
<synopsis>The address and port of the Homer server to send packets to.</synopsis>
</configOption>
<configOption name="capture_password">
<synopsis>If set, the authentication password to send to Homer.</synopsis>
</configOption>
<configOption name="capture_id" default="0">
<synopsis>The ID for this capture agent.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_statsd" language="en_US">
<synopsis>Statsd client.</synopsis>
<configFile name="statsd.conf">
<configObject name="global">
<synopsis>Global configuration settings</synopsis>
<configOption name="enabled">
<synopsis>Enable/disable the statsd module</synopsis>
</configOption>
<configOption name="server">
<synopsis>Address of the statsd server</synopsis>
</configOption>
<configOption name="prefix">
<synopsis>Prefix to prepend to every metric</synopsis>
</configOption>
<configOption name="add_newline">
<synopsis>Append a newline to every event. This is useful if you want to fake out a server using netcat (nc -lu 8125)</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_pjsip_outbound_publish" language="en_US">
<synopsis>SIP resource for outbound publish</synopsis>
<description>
<para>
<emphasis>Outbound Publish</emphasis>
</para>
<para>This module allows <literal>res_pjsip</literal> to publish to other SIP servers.</para>
</description>
<configFile name="pjsip.conf">
<configObject name="outbound-publish">
<synopsis>The configuration for outbound publish</synopsis>
<description>
<para>
Publish is <emphasis>COMPLETELY</emphasis> separate from the rest of
<literal>pjsip.conf</literal>. A minimal configuration consists of
setting a <literal>server_uri</literal> and <literal>event</literal>.
</para>
</description>
<configOption name="expiration" default="3600">
<synopsis>Expiration time for publications in seconds</synopsis>
</configOption>
<configOption name="outbound_auth" default="">
<synopsis>Authentication object(s) to be used for outbound publishes.</synopsis>
<description>
<para>
This is a comma-delimited list of <replaceable>auth</replaceable>
sections defined in <filename>pjsip.conf</filename> used to respond
to outbound authentication challenges.</para>
<note>
<para>
Using the same auth section for inbound and outbound
authentication is not recommended. There is a difference in
meaning for an empty realm setting between inbound and outbound
authentication uses. See the auth realm description for details.
</para>
</note>
</description>
</configOption>
<configOption name="outbound_proxy" default="">
<synopsis>Full SIP URI of the outbound proxy used to send publishes</synopsis>
</configOption>
<configOption name="server_uri">
<synopsis>SIP URI of the server and entity to publish to</synopsis>
<description>
<para>
This is the URI at which to find the entity and server to send the outbound PUBLISH to.
This URI is used as the request URI of the outbound PUBLISH request from Asterisk.
</para>
</description>
</configOption>
<configOption name="from_uri">
<synopsis>SIP URI to use in the From header</synopsis>
<description>
<para>
This is the URI that will be placed into the From header of outgoing PUBLISH
messages. If no URI is specified then the URI provided in <literal>server_uri</literal>
will be used.
</para>
</description>
</configOption>
<configOption name="to_uri">
<synopsis>SIP URI to use in the To header</synopsis>
<description>
<para>
This is the URI that will be placed into the To header of outgoing PUBLISH
messages. If no URI is specified then the URI provided in <literal>server_uri</literal>
will be used.
</para>
</description>
</configOption>
<configOption name="event" default="">
<synopsis>Event type of the PUBLISH.</synopsis>
</configOption>
<configOption name="max_auth_attempts" default="5">
<synopsis>Maximum number of authentication attempts before stopping the publication.</synopsis>
</configOption>
<configOption name="transport">
<synopsis>Transport used for outbound publish</synopsis>
<description>
<note>
<para>A <replaceable>transport</replaceable> configured in
<literal>pjsip.conf</literal>. As with other <literal>res_pjsip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para>
</note>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'outbound-publish'.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_mwi_external" language="en_US">
<synopsis>Core external MWI support</synopsis>
<configFile name="sorcery.conf">
<configObject name="mailboxes">
<synopsis>Persistent cache of external MWI Mailboxs.</synopsis>
<description>
<para>Allows the alteration of sorcery backend mapping for
the persistent cache of external MWI mailboxes.</para>
</description>
</configObject>
</configFile>
</configInfo>
<function name="CALENDAR_BUSY" language="en_US">
<synopsis>
Determine if the calendar is marked busy at this time.
</synopsis>
<syntax>
<parameter name="calendar" required="true" />
</syntax>
<description>
<para>Check the specified calendar's current busy status.</para>
</description>
<see-also>
<ref type="function">CALENDAR_EVENT</ref>
<ref type="function">CALENDAR_QUERY</ref>
<ref type="function">CALENDAR_QUERY_RESULT</ref>
<ref type="function">CALENDAR_WRITE</ref>
</see-also>
</function>
<function name="CALENDAR_EVENT" language="en_US">
<synopsis>
Get calendar event notification data from a notification call.
</synopsis>
<syntax>
<parameter name="field" required="true">
<enumlist>
<enum name="summary">
<para>The VEVENT SUMMARY property or Exchange event 'subject'</para>
</enum>
<enum name="description">
<para>The text description of the event</para>
</enum>
<enum name="organizer">
<para>The organizer of the event</para>
</enum>
<enum name="location">
<para>The location of the eventt</para>
</enum>
<enum name="categories">
<para>The categories of the event</para>
</enum>
<enum name="priority">
<para>The priority of the event</para>
</enum>
<enum name="calendar">
<para>The name of the calendar associated with the event</para>
</enum>
<enum name="uid">
<para>The unique identifier for this event</para>
</enum>
<enum name="start">
<para>The start time of the event</para>
</enum>
<enum name="end">
<para>The end time of the event</para>
</enum>
<enum name="busystate">
<para>The busy state of the event 0=FREE, 1=TENTATIVE, 2=BUSY</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Whenever a calendar event notification call is made, the event data
may be accessed with this function.</para>
</description>
<see-also>
<ref type="function">CALENDAR_BUSY</ref>
<ref type="function">CALENDAR_QUERY</ref>
<ref type="function">CALENDAR_QUERY_RESULT</ref>
<ref type="function">CALENDAR_WRITE</ref>
</see-also>
</function>
<function name="CALENDAR_QUERY" language="en_US">
<synopsis>Query a calendar server and store the data on a channel
</synopsis>
<syntax>
<parameter name="calendar" required="true">
<para>The calendar that should be queried</para>
</parameter>
<parameter name="start" required="false">
<para>The start time of the query (in seconds since epoch)</para>
</parameter>
<parameter name="end" required="false">
<para>The end time of the query (in seconds since epoch)</para>
</parameter>
</syntax>
<description>
<para>Get a list of events in the currently accessible timeframe of the <replaceable>calendar</replaceable>
The function returns the id for accessing the result with CALENDAR_QUERY_RESULT()</para>
</description>
<see-also>
<ref type="function">CALENDAR_BUSY</ref>
<ref type="function">CALENDAR_EVENT</ref>
<ref type="function">CALENDAR_QUERY_RESULT</ref>
<ref type="function">CALENDAR_WRITE</ref>
</see-also>
</function>
<function name="CALENDAR_QUERY_RESULT" language="en_US">
<synopsis>
Retrieve data from a previously run CALENDAR_QUERY() call
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The query ID returned by <literal>CALENDAR_QUERY</literal></para>
</parameter>
<parameter name="field" required="true">
<enumlist>
<enum name="getnum">
<para>number of events occurring during time range</para>
</enum>
<enum name="summary">
<para>A summary of the event</para>
</enum>
<enum name="description">
<para>The full event description</para>
</enum>
<enum name="organizer">
<para>The event organizer</para>
</enum>
<enum name="location">
<para>The event location</para>
</enum>
<enum name="categories">
<para>The categories of the event</para>
</enum>
<enum name="priority">
<para>The priority of the event</para>
</enum>
<enum name="calendar">
<para>The name of the calendar associted with the event</para>
</enum>
<enum name="uid">
<para>The unique identifier for the event</para>
</enum>
<enum name="start">
<para>The start time of the event (in seconds since epoch)</para>
</enum>
<enum name="end">
<para>The end time of the event (in seconds since epoch)</para>
</enum>
<enum name="busystate">
<para>The busy status of the event 0=FREE, 1=TENTATIVE, 2=BUSY</para>
</enum>
</enumlist>
</parameter>
<parameter name="entry" required="false" default="1">
<para>Return data from a specific event returned by the query</para>
</parameter>
</syntax>
<description>
<para>After running CALENDAR_QUERY and getting a result <replaceable>id</replaceable>, calling
<literal>CALENDAR_QUERY</literal> with that <replaceable>id</replaceable> and a <replaceable>field</replaceable>
will return the data for that field. If multiple events matched the query, and <replaceable>entry</replaceable>
is provided, information from that event will be returned.</para>
</description>
<see-also>
<ref type="function">CALENDAR_BUSY</ref>
<ref type="function">CALENDAR_EVENT</ref>
<ref type="function">CALENDAR_QUERY</ref>
<ref type="function">CALENDAR_WRITE</ref>
</see-also>
</function>
<function name="CALENDAR_WRITE" language="en_US">
<synopsis>Write an event to a calendar</synopsis>
<syntax>
<parameter name="calendar" required="true">
<para>The calendar to write to</para>
</parameter>
<parameter name="field" multiple="true" required="true">
<enumlist>
<enum name="summary">
<para>A summary of the event</para>
</enum>
<enum name="description">
<para>The full event description</para>
</enum>
<enum name="organizer">
<para>The event organizer</para>
</enum>
<enum name="location">
<para>The event location</para>
</enum>
<enum name="categories">
<para>The categories of the event</para>
</enum>
<enum name="priority">
<para>The priority of the event</para>
</enum>
<enum name="uid">
<para>The unique identifier for the event</para>
</enum>
<enum name="start">
<para>The start time of the event (in seconds since epoch)</para>
</enum>
<enum name="end">
<para>The end time of the event (in seconds since epoch)</para>
</enum>
<enum name="busystate">
<para>The busy status of the event 0=FREE, 1=TENTATIVE, 2=BUSY</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Example: CALENDAR_WRITE(calendar,field1,field2,field3)=val1,val2,val3</para>
<para>The field and value arguments can easily be set/passed using the HASHKEYS() and HASH() functions</para>
<variablelist>
<variable name="CALENDAR_SUCCESS">
<para>The status of the write operation to the calendar</para>
<value name="1" >
The event was successfully written to the calendar.
</value>
<value name="0" >
The event was not written to the calendar due to network issues, permissions, etc.
</value>
</variable>
</variablelist>
</description>
<see-also>
<ref type="function">CALENDAR_BUSY</ref>
<ref type="function">CALENDAR_EVENT</ref>
<ref type="function">CALENDAR_QUERY</ref>
<ref type="function">CALENDAR_QUERY_RESULT</ref>
</see-also>
</function>
<function name="SMDI_MSG_RETRIEVE" language="en_US">
<synopsis>
Retrieve an SMDI message.
</synopsis>
<syntax>
<parameter name="smdi port" required="true" />
<parameter name="search key" required="true" />
<parameter name="timeout" />
<parameter name="options">
<enumlist>
<enum name="t">
<para>Instead of searching on the forwarding station, search on the message desk terminal.</para>
</enum>
<enum name="n">
<para>Instead of searching on the forwarding station, search on the message desk number.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function is used to retrieve an incoming SMDI message. It returns
an ID which can be used with the SMDI_MSG() function to access details of
the message. Note that this is a destructive function in the sense that
once an SMDI message is retrieved using this function, it is no longer in
the global SMDI message queue, and can not be accessed by any other Asterisk
channels. The timeout for this function is optional, and the default is
3 seconds. When providing a timeout, it should be in milliseconds.
</para>
<para>The default search is done on the forwarding station ID. However, if
you set one of the search key options in the options field, you can change
this behavior.
</para>
</description>
<see-also>
<ref type="function">SMDI_MSG</ref>
</see-also>
</function>
<function name="SMDI_MSG" language="en_US">
<synopsis>
Retrieve details about an SMDI message.
</synopsis>
<syntax>
<parameter name="message_id" required="true" />
<parameter name="component" required="true">
<para>Valid message components are:</para>
<enumlist>
<enum name="number">
<para>The message desk number</para>
</enum>
<enum name="terminal">
<para>The message desk terminal</para>
</enum>
<enum name="station">
<para>The forwarding station</para>
</enum>
<enum name="callerid">
<para>The callerID of the calling party that was forwarded</para>
</enum>
<enum name="type">
<para>The call type. The value here is the exact character
that came in on the SMDI link. Typically, example values
are:</para>
<para>Options:</para>
<enumlist>
<enum name="D">
<para>Direct Calls</para>
</enum>
<enum name="A">
<para>Forward All Calls</para>
</enum>
<enum name="B">
<para>Forward Busy Calls</para>
</enum>
<enum name="N">
<para>Forward No Answer Calls</para>
</enum>
</enumlist>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function is used to access details of an SMDI message that was
pulled from the incoming SMDI message queue using the SMDI_MSG_RETRIEVE()
function.</para>
</description>
<see-also>
<ref type="function">SMDI_MSG_RETRIEVE</ref>
</see-also>
</function>
<agi name="answer" language="en_US">
<synopsis>
Answer channel
</synopsis>
<syntax />
<description>
<para>Answers channel if not already in answer state. Returns <literal>-1</literal> on
channel failure, or <literal>0</literal> if successful.</para>
</description>
<see-also>
<ref type="agi">hangup</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="asyncagi break" language="en_US">
<synopsis>
Interrupts Async AGI
</synopsis>
<syntax />
<description>
<para>Interrupts expected flow of Async AGI commands and returns control to previous source
(typically, the PBX dialplan).</para>
</description>
<see-also>
<ref type="agi">hangup</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="channel status" language="en_US">
<synopsis>
Returns status of the connected channel.
</synopsis>
<syntax>
<parameter name="channelname" />
</syntax>
<description>
<para>Returns the status of the specified <replaceable>channelname</replaceable>.
If no channel name is given then returns the status of the current channel.</para>
<para>Return values:</para>
<enumlist>
<enum name="0">
<para>Channel is down and available.</para>
</enum>
<enum name="1">
<para>Channel is down, but reserved.</para>
</enum>
<enum name="2">
<para>Channel is off hook.</para>
</enum>
<enum name="3">
<para>Digits (or equivalent) have been dialed.</para>
</enum>
<enum name="4">
<para>Line is ringing.</para>
</enum>
<enum name="5">
<para>Remote end is ringing.</para>
</enum>
<enum name="6">
<para>Line is up.</para>
</enum>
<enum name="7">
<para>Line is busy.</para>
</enum>
</enumlist>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="control stream file" language="en_US">
<synopsis>
Sends audio file on channel and allows the listener to control the stream.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>The file extension must not be included in the filename.</para>
</parameter>
<parameter name="escape_digits" required="true" />
<parameter name="skipms" />
<parameter name="ffchar">
<para>Defaults to <literal>#</literal></para>
</parameter>
<parameter name="rewchr">
<para>Defaults to <literal>*</literal></para>
</parameter>
<parameter name="pausechr" />
<parameter name="offsetms">
<para>Offset, in milliseconds, to start the audio playback</para>
</parameter>
</syntax>
<description>
<para>Send the given file, allowing playback to be controlled by the given
digits, if any. Use double quotes for the digits if you wish none to be
permitted. If offsetms is provided then the audio will seek to offsetms
before play starts. Returns <literal>0</literal> if playback completes without a digit
being pressed, or the ASCII numerical value of the digit if one was pressed,
or <literal>-1</literal> on error or if the channel was disconnected. Returns the
position where playback was terminated as endpos.</para>
<para>It sets the following channel variables upon completion:</para>
<variablelist>
<variable name="CPLAYBACKSTATUS">
<para>Contains the status of the attempt as a text string</para>
<value name="SUCCESS" />
<value name="USERSTOPPED" />
<value name="REMOTESTOPPED" />
<value name="ERROR" />
</variable>
<variable name="CPLAYBACKOFFSET">
<para>Contains the offset in ms into the file where playback
was at when it stopped. <literal>-1</literal> is end of file.</para>
</variable>
<variable name="CPLAYBACKSTOPKEY">
<para>If the playback is stopped by the user this variable contains
the key that was pressed.</para>
</variable>
</variablelist>
</description>
<see-also>
<ref type="agi">get option</ref>
<ref type="agi">control stream file</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database del" language="en_US">
<synopsis>
Removes database key/value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>Deletes an entry in the Asterisk database for a given
<replaceable>family</replaceable> and <replaceable>key</replaceable>.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal>
otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database put</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database deltree" language="en_US">
<synopsis>
Removes database keytree/value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="keytree" />
</syntax>
<description>
<para>Deletes a <replaceable>family</replaceable> or specific <replaceable>keytree</replaceable>
within a <replaceable>family</replaceable> in the Asterisk database.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal> otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database put</ref>
<ref type="agi">database del</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database get" language="en_US">
<synopsis>
Gets database value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>Retrieves an entry in the Asterisk database for a given <replaceable>family</replaceable>
and <replaceable>key</replaceable>.</para>
<para>Returns <literal>0</literal> if <replaceable>key</replaceable> is not set.
Returns <literal>1</literal> if <replaceable>key</replaceable> is set and returns the variable
in parenthesis.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">database put</ref>
<ref type="agi">database del</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database put" language="en_US">
<synopsis>
Adds/updates database value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Adds or updates an entry in the Asterisk database for a given
<replaceable>family</replaceable>, <replaceable>key</replaceable>, and
<replaceable>value</replaceable>.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal> otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database del</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="exec" language="en_US">
<synopsis>
Executes a given Application
</synopsis>
<syntax>
<parameter name="application" required="true" />
<parameter name="options" required="true" />
</syntax>
<description>
<para>Executes <replaceable>application</replaceable> with given
<replaceable>options</replaceable>.</para>
<para>Returns whatever the <replaceable>application</replaceable> returns, or
<literal>-2</literal> on failure to find <replaceable>application</replaceable>.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get data" language="en_US">
<synopsis>
Prompts for DTMF on a channel
</synopsis>
<syntax>
<parameter name="file" required="true" />
<parameter name="timeout" />
<parameter name="maxdigits" />
</syntax>
<description>
<para>Stream the given <replaceable>file</replaceable>, and receive DTMF data.</para>
<para>Returns the digits received from the channel at the other end.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get full variable" language="en_US">
<synopsis>
Evaluates a channel expression
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
<parameter name="channel name" />
</syntax>
<description>
<para>Returns <literal>0</literal> if <replaceable>variablename</replaceable> is not set
or channel does not exist. Returns <literal>1</literal> if <replaceable>variablename</replaceable>
is set and returns the variable in parenthesis. Understands complex variable names and builtin
variables, unlike GET VARIABLE.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">get variable</ref>
<ref type="agi">set variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get option" language="en_US">
<synopsis>
Stream file, prompt for DTMF, with timeout.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="escape_digits" required="true" />
<parameter name="timeout" />
</syntax>
<description>
<para>Behaves similar to STREAM FILE but used with a timeout option.</para>
</description>
<see-also>
<ref type="agi">stream file</ref>
<ref type="agi">control stream file</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get variable" language="en_US">
<synopsis>
Gets a channel variable.
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
</syntax>
<description>
<para>Returns <literal>0</literal> if <replaceable>variablename</replaceable> is not set.
Returns <literal>1</literal> if <replaceable>variablename</replaceable> is set and returns
the variable in parentheses.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">get full variable</ref>
<ref type="agi">set variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="hangup" language="en_US">
<synopsis>
Hangup a channel.
</synopsis>
<syntax>
<parameter name="channelname" />
</syntax>
<description>
<para>Hangs up the specified channel. If no channel name is given, hangs
up the current channel</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="noop" language="en_US">
<synopsis>
Does nothing.
</synopsis>
<syntax />
<description>
<para>Does nothing.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="receive char" language="en_US">
<synopsis>
Receives one character from channels supporting it.
</synopsis>
<syntax>
<parameter name="timeout" required="true">
<para>The maximum time to wait for input in milliseconds, or <literal>0</literal>
for infinite. Most channels</para>
</parameter>
</syntax>
<description>
<para>Receives a character of text on a channel. Most channels do not support
the reception of text. Returns the decimal value of the character
if one is received, or <literal>0</literal> if the channel does not support
text reception. Returns <literal>-1</literal> only on error/hangup.</para>
</description>
<see-also>
<ref type="agi">receive text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="receive text" language="en_US">
<synopsis>
Receives text from channels supporting it.
</synopsis>
<syntax>
<parameter name="timeout" required="true">
<para>The timeout to be the maximum time to wait for input in
milliseconds, or <literal>0</literal> for infinite.</para>
</parameter>
</syntax>
<description>
<para>Receives a string of text on a channel. Most channels
do not support the reception of text. Returns <literal>-1</literal> for failure
or <literal>1</literal> for success, and the string in parenthesis.</para>
</description>
<see-also>
<ref type="agi">receive char</ref>
<ref type="agi">send text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="record file" language="en_US">
<synopsis>
Records to a given file.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>The destination filename of the recorded audio.</para>
</parameter>
<parameter name="format" required="true">
<para>The audio format in which to save the resulting file.</para>
</parameter>
<parameter name="escape_digits" required="true">
<para>The DTMF digits that will terminate the recording process.</para>
</parameter>
<parameter name="timeout" required="true">
<para>The maximum recording time in milliseconds. Set to -1 for no
limit.</para>
</parameter>
<parameter name="offset_samples">
<para>Causes the recording to first seek to the specified offset before
recording begins.</para>
</parameter>
<parameter name="beep">
<para>Causes Asterisk to play a beep as recording begins. This argument
can take any value.</para>
</parameter>
<parameter name="s=silence">
<para>The number of seconds of silence that are permitted before the
recording is terminated, regardless of the
<replaceable>escape_digits</replaceable> or <replaceable>timeout</replaceable>
arguments. If specified, this parameter must be preceded by
<literal>s=</literal>.</para>
</parameter>
</syntax>
<description>
<para>Record to a file until a given dtmf digit in the sequence is received.
Returns <literal>-1</literal> on hangup or error. The format will specify what kind of file
will be recorded. The <replaceable>timeout</replaceable> is the maximum record time in
milliseconds, or <literal>-1</literal> for no <replaceable>timeout</replaceable>.
<replaceable>offset samples</replaceable> is optional, and, if provided, will seek
to the offset without exceeding the end of the
file. <replaceable>beep</replaceable> can take any value, and causes Asterisk
to play a beep to the channel that is about to be recorded. <replaceable>silence</replaceable> is
the number of seconds of silence allowed before the function returns despite the
lack of dtmf digits or reaching <replaceable>timeout</replaceable>. <replaceable>silence</replaceable>
value must be preceded by <literal>s=</literal> and is also optional.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say alpha" language="en_US">
<synopsis>
Says a given character string.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given character string, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if one
was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say digits" language="en_US">
<synopsis>
Says a given digit string.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given digit string, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if one
was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say number" language="en_US">
<synopsis>
Says a given number.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
<parameter name="gender" />
</syntax>
<description>
<para>Say a given number, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of
the digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say phonetic" language="en_US">
<synopsis>
Says a given character string with phonetics.
</synopsis>
<syntax>
<parameter name="string" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given character string with phonetics, returning early if any of the
given DTMF digits are received on the channel. Returns <literal>0</literal> if
playback completes without a digit pressed, the ASCII numerical value of the digit
if one was pressed, or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say date" language="en_US">
<synopsis>
Says a given date.
</synopsis>
<syntax>
<parameter name="date" required="true">
<para>Is number of seconds elapsed since 00:00:00 on January 1, 1970.
Coordinated Universal Time (UTC).</para>
</parameter>
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given date, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of the
digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say time" language="en_US">
<synopsis>
Says a given time.
</synopsis>
<syntax>
<parameter name="time" required="true">
<para>Is number of seconds elapsed since 00:00:00 on January 1, 1970.
Coordinated Universal Time (UTC).</para>
</parameter>
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given time, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if
one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say datetime" language="en_US">
<synopsis>
Says a given time as specified by the format given.
</synopsis>
<syntax>
<parameter name="time" required="true">
<para>Is number of seconds elapsed since 00:00:00
on January 1, 1970, Coordinated Universal Time (UTC)</para>
</parameter>
<parameter name="escape_digits" required="true" />
<parameter name="format">
<para>Is the format the time should be said in. See
<filename>voicemail.conf</filename> (defaults to <literal>ABdY
'digits/at' IMp</literal>).</para>
</parameter>
<parameter name="timezone">
<para>Acceptable values can be found in <filename>/usr/share/zoneinfo</filename>
Defaults to machine default.</para>
</parameter>
</syntax>
<description>
<para>Say a given time, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of the
digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="send image" language="en_US">
<synopsis>
Sends images to channels supporting it.
</synopsis>
<syntax>
<parameter name="image" required="true" />
</syntax>
<description>
<para>Sends the given image on a channel. Most channels do not support the
transmission of images. Returns <literal>0</literal> if image is sent, or if
the channel does not support image transmission. Returns <literal>-1</literal>
only on error/hangup. Image names should not include extensions.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="send text" language="en_US">
<synopsis>
Sends text to channels supporting it.
</synopsis>
<syntax>
<parameter name="text to send" required="true">
<para>Text consisting of greater than one word should be placed
in quotes since the command only accepts a single argument.</para>
</parameter>
</syntax>
<description>
<para>Sends the given text on a channel. Most channels do not support the
transmission of text. Returns <literal>0</literal> if text is sent, or if the
channel does not support text transmission. Returns <literal>-1</literal> only
on error/hangup.</para>
</description>
<see-also>
<ref type="agi">receive text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set autohangup" language="en_US">
<synopsis>
Autohangup channel in some time.
</synopsis>
<syntax>
<parameter name="time" required="true" />
</syntax>
<description>
<para>Cause the channel to automatically hangup at <replaceable>time</replaceable>
seconds in the future. Of course it can be hungup before then as well. Setting to
<literal>0</literal> will cause the autohangup feature to be disabled on this channel.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set callerid" language="en_US">
<synopsis>
Sets callerid for the current channel.
</synopsis>
<syntax>
<parameter name="number" required="true" />
</syntax>
<description>
<para>Changes the callerid of the current channel.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set context" language="en_US">
<synopsis>
Sets channel context.
</synopsis>
<syntax>
<parameter name="desired context" required="true" />
</syntax>
<description>
<para>Sets the context for continuation upon exiting the application.</para>
</description>
<see-also>
<ref type="agi">set extension</ref>
<ref type="agi">set priority</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set extension" language="en_US">
<synopsis>
Changes channel extension.
</synopsis>
<syntax>
<parameter name="new extension" required="true" />
</syntax>
<description>
<para>Changes the extension for continuation upon exiting the application.</para>
</description>
<see-also>
<ref type="agi">set context</ref>
<ref type="agi">set priority</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set music" language="en_US">
<synopsis>
Enable/Disable Music on hold generator
</synopsis>
<syntax>
<parameter required="true">
<enumlist>
<enum>
<parameter name="on" literal="true" required="true" />
</enum>
<enum>
<parameter name="off" literal="true" required="true" />
</enum>
</enumlist>
</parameter>
<parameter name="class" required="true" />
</syntax>
<description>
<para>Enables/Disables the music on hold generator. If <replaceable>class</replaceable>
is not specified, then the <literal>default</literal> music on hold class will be
used. This generator will be stopped automatically when playing a file.</para>
<para>Always returns <literal>0</literal>.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set priority" language="en_US">
<synopsis>
Set channel dialplan priority.
</synopsis>
<syntax>
<parameter name="priority" required="true" />
</syntax>
<description>
<para>Changes the priority for continuation upon exiting the application.
The priority must be a valid priority or label.</para>
</description>
<see-also>
<ref type="agi">set context</ref>
<ref type="agi">set extension</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set variable" language="en_US">
<synopsis>
Sets a channel variable.
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Sets a variable to the current channel.</para>
</description>
<see-also>
<ref type="agi">get variable</ref>
<ref type="agi">get full variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="stream file" language="en_US">
<synopsis>
Sends audio file on channel.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>File name to play. The file extension must not be
included in the <replaceable>filename</replaceable>.</para>
</parameter>
<parameter name="escape_digits" required="true">
<para>Use double quotes for the digits if you wish none to be
permitted.</para>
</parameter>
<parameter name="sample offset">
<para>If sample offset is provided then the audio will seek to sample
offset before play starts.</para>
</parameter>
</syntax>
<description>
<para>Send the given file, allowing playback to be interrupted by the given
digits, if any. Returns <literal>0</literal> if playback completes without a digit
being pressed, or the ASCII numerical value of the digit if one was pressed,
or <literal>-1</literal> on error or if the channel was disconnected. If
musiconhold is playing before calling stream file it will be automatically
stopped and will not be restarted after completion.</para>
<para>It sets the following channel variables upon completion:</para>
<variablelist>
<variable name="PLAYBACKSTATUS">
<para>The status of the playback attempt as a text string.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
</description>
<see-also>
<ref type="agi">control stream file</ref>
<ref type="agi">get option</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="tdd mode" language="en_US">
<synopsis>
Toggles TDD mode (for the deaf).
</synopsis>
<syntax>
<parameter name="boolean" required="true">
<enumlist>
<enum name="on" />
<enum name="off" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Enable/Disable TDD transmission/reception on a channel. Returns <literal>1</literal> if
successful, or <literal>0</literal> if channel is not TDD-capable.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="verbose" language="en_US">
<synopsis>
Logs a message to the asterisk verbose log.
</synopsis>
<syntax>
<parameter name="message" required="true" />
<parameter name="level" required="true" />
</syntax>
<description>
<para>Sends <replaceable>message</replaceable> to the console via verbose
message system. <replaceable>level</replaceable> is the verbose level (1-4).
Always returns <literal>1</literal>
</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="wait for digit" language="en_US">
<synopsis>
Waits for a digit to be pressed.
</synopsis>
<syntax>
<parameter name="timeout" required="true" />
</syntax>
<description>
<para>Waits up to <replaceable>timeout</replaceable> milliseconds for channel to
receive a DTMF digit. Returns <literal>-1</literal> on channel failure, <literal>0</literal>
if no digit is received in the timeout, or the numerical value of the ascii of the digit if
one is received. Use <literal>-1</literal> for the <replaceable>timeout</replaceable> value if
you desire the call to block indefinitely.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech create" language="en_US">
<synopsis>
Creates a speech object.
</synopsis>
<syntax>
<parameter name="engine" required="true" />
</syntax>
<description>
<para>Create a speech object to be used by the other Speech AGI commands.</para>
</description>
<see-also>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech set" language="en_US">
<synopsis>
Sets a speech engine setting.
</synopsis>
<syntax>
<parameter name="name" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Set an engine-specific setting.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech destroy" language="en_US">
<synopsis>
Destroys a speech object.
</synopsis>
<syntax>
</syntax>
<description>
<para>Destroy the speech object created by <literal>SPEECH CREATE</literal>.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech load grammar" language="en_US">
<synopsis>
Loads a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
<parameter name="path to grammar" required="true" />
</syntax>
<description>
<para>Loads the specified grammar as the specified name.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech unload grammar" language="en_US">
<synopsis>
Unloads a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Unloads the specified grammar.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech activate grammar" language="en_US">
<synopsis>
Activates a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Activates the specified grammar on the speech object.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech deactivate grammar" language="en_US">
<synopsis>
Deactivates a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Deactivates the specified grammar on the speech object.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech recognize" language="en_US">
<synopsis>
Recognizes speech.
</synopsis>
<syntax>
<parameter name="prompt" required="true" />
<parameter name="timeout" required="true" />
<parameter name="offset" />
</syntax>
<description>
<para>Plays back given <replaceable>prompt</replaceable> while listening for
speech and dtmf.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<application name="AGI" language="en_US">
<synopsis>
Executes an AGI compliant application.
</synopsis>
<syntax>
<parameter name="command" required="true">
<para>How AGI should be invoked on the channel.</para>
</parameter>
<parameter name="args">
<para>Arguments to pass to the AGI script or server.</para>
<argument name="arg1" required="true" />
<argument name="arg2" multiple="yes" />
</parameter>
</syntax>
<description>
<para>Executes an Asterisk Gateway Interface compliant
program on a channel. AGI allows Asterisk to launch external programs written
in any language to control a telephony channel, play audio, read DTMF digits,
etc. by communicating with the AGI protocol.</para>
<para>The following variants of AGI exist, and are chosen based on the value
passed to <replaceable>command</replaceable>:</para>
<enumlist>
<enum name="AGI">
<para>The classic variant of AGI, this will launch the script
specified by <replaceable>command</replaceable> as a new process.
Communication with the script occurs on <literal>stdin</literal> and
<literal>stdout</literal>. If the full path to the script is not
provided, the <directory>astagidir</directory> specified in
<filename>asterisk.conf</filename> will be used.
</para>
</enum>
<enum name="FastAGI">
<para>Connect Asterisk to a FastAGI server using a TCP connection.
The URI to the FastAGI server should be given in the form
<literal>[scheme]://host.domain[:port][/script/name]</literal>,
where <replaceable>scheme</replaceable> is either <literal>agi</literal>
or <literal>hagi</literal>.</para>
<para>In the case of <literal>hagi</literal>, an SRV lookup will be
performed to try to connect to a list of FastAGI servers. The hostname in
the URI must be prefixed with <literal>_agi._tcp</literal>. prior to the DNS resolution. For
example, if you specify the URI <literal>hagi://agi.example.com/foo.agi</literal>
the DNS query would be for <literal>_agi._tcp.agi.example.com</literal>. You
will need to make sure this resolves correctly.</para>
</enum>
<enum name="AsyncAGI">
<para>Use AMI to control the channel in AGI. AGI commands can be invoked
using the <literal>AMI</literal> action, with a variety of AGI specific
events passed back over the AMI connection. AsyncAGI should be invoked
by passing <literal>agi:async</literal> to the <replaceable>command</replaceable>
parameter.</para>
</enum>
</enumlist>
<note>
<para>As of <literal>1.6.0</literal>, this channel will
not stop dialplan execution on hangup inside of this application. Dialplan
execution will continue normally, even upon hangup until the AGI application
signals a desire to stop (either by exiting or, in the case of a net script, by
closing the connection).</para>
<para>A locally executed AGI script will receive <literal>SIGHUP</literal> on
hangup from the channel except when using <literal>DeadAGI</literal>
(or when the channel is already hungup). A fast AGI server will
correspondingly receive a <literal>HANGUP</literal> inline with the command dialog.
Both of these signals may be disabled by setting the <variable>AGISIGHUP</variable>
channel variable to <literal>no</literal> before executing the AGI application.
Alternatively, if you would like the AGI application to exit immediately
after a channel hangup is detected, set the <variable>AGIEXITONHANGUP</variable>
variable to <literal>yes</literal>.</para>
</note>
<example title="AGI invocation examples">
; Start the AGI script /tmp/my-cool-script.sh, passing it the contents
; of the channel variable FOO
same => n,AGI(/tmp/my-cool-script.sh,${FOO})
; Start the AGI script my-cool-script.sh located in the astagidir
; directory, specified in asterisk.conf
same => n,AGI(my-cool-script.sh)
; Connect to the FastAGI server located at 127.0.0.1 and start the script
; awesome-script
same => n,AGI(agi://127.0.0.1/awesome-script)
; Start AsyncAGI
same => n,AGI(agi:async)
</example>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="AGISTATUS">
<para>The status of the attempt to the run the AGI script
text string, one of:</para>
<value name="SUCCESS" />
<value name="FAILURE" />
<value name="NOTFOUND" />
<value name="HANGUP" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="manager">AGI</ref>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="application">EAGI</ref>
<ref type="application">DeadAGI</ref>
<ref type="filename">asterisk.conf</ref>
</see-also>
</application>
<application name="EAGI" language="en_US">
<synopsis>
Executes an EAGI compliant application.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='command'])" />
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='args'])" />
</syntax>
<description>
<para>Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
on file descriptor 3. In all other respects, it behaves in the same fashion as
AGI. See the documentation for the <literal>AGI</literal> dialplan application for
more information on invoking AGI on a channel.</para>
<para>This application sets the following channel variable upon completion:</para>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/description/variablelist)" />
</description>
<see-also>
<ref type="application">AGI</ref>
<ref type="application">DeadAGI</ref>
</see-also>
</application>
<application name="DeadAGI" language="en_US">
<synopsis>
Executes AGI on a hungup channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='command'])" />
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='args'])" />
</syntax>
<description>
<warning>
<para>This application is deprecated and may be removed in a future version
of Asterisk. Use the replacement application <literal>AGI</literal> instead
of <literal>DeadAGI</literal>.
</para>
</warning>
<para>Execute AGI on a 'dead' or hungup channel. See the documentation for the
<literal>AGI</literal> dialplan application for more information on invoking
AGI on a channel.</para>
<para>This application sets the following channel variable upon completion:</para>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/description/variablelist)" />
</description>
<see-also>
<ref type="application">AGI</ref>
<ref type="application">EAGI</ref>
</see-also>
</application>
<manager name="AGI" language="en_US">
<synopsis>
Add an AGI command to execute by Async AGI.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel that is currently in Async AGI.</para>
</parameter>
<parameter name="Command" required="true">
<para>Application to execute.</para>
</parameter>
<parameter name="CommandID">
<para>This will be sent back in CommandID header of AsyncAGI exec
event notification.</para>
</parameter>
</syntax>
<description>
<para>Add an AGI command to the execute queue of the channel in Async AGI.</para>
</description>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="AsyncAGIStart">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a channel starts AsyncAGI command processing.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Env">
<para>URL encoded string read from the AsyncAGI server.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AsyncAGIEnd">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a channel stops AsyncAGI command processing.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AsyncAGIExec">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when AsyncAGI completes an AGI command.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="CommandID" required="false">
<para>Optional command ID sent by the AsyncAGI server to identify the command.</para>
</parameter>
<parameter name="Result">
<para>URL encoded result string from the executed AGI command.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AGIExecStart">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a received AGI command starts processing.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Command">
<para>The AGI command as received from the external source.</para>
</parameter>
<parameter name="CommandId">
<para>Random identification number assigned to the execution of this command.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AGIExecEnd</ref>
<ref type="application">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AGIExecEnd">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a received AGI command completes processing.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AGIExecStart']/managerEventInstance/syntax/parameter)" />
<parameter name="ResultCode">
<para>The numeric result code from AGI</para>
</parameter>
<parameter name="Result">
<para>The text result reason from AGI</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AGIExecStart</ref>
<ref type="application">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="PJSIPShowSubscriptionsInbound" language="en_US">
<synopsis>
Lists subscriptions.
</synopsis>
<syntax />
<description>
<para>
Provides a listing of all inbound subscriptions. An event <literal>InboundSubscriptionDetail</literal>
is issued for each subscription object. Once all detail events are completed an
<literal>InboundSubscriptionDetailComplete</literal> event is issued.
</para>
</description>
</manager>
<manager name="PJSIPShowSubscriptionsOutbound" language="en_US">
<synopsis>
Lists subscriptions.
</synopsis>
<syntax />
<description>
<para>
Provides a listing of all outbound subscriptions. An event <literal>OutboundSubscriptionDetail</literal>
is issued for each subscription object. Once all detail events are completed an
<literal>OutboundSubscriptionDetailComplete</literal> event is issued.
</para>
</description>
</manager>
<manager name="PJSIPShowResourceLists" language="en_US">
<synopsis>
Displays settings for configured resource lists.
</synopsis>
<syntax />
<description>
<para>
Provides a listing of all resource lists. An event <literal>ResourceListDetail</literal>
is issued for each resource list object. Once all detail events are completed a
<literal>ResourceListDetailComplete</literal> event is issued.
</para>
</description>
</manager>
<configInfo name="res_pjsip_pubsub" language="en_US">
<synopsis>Module that implements publish and subscribe support.</synopsis>
<configFile name="pjsip.conf">
<configObject name="subscription_persistence">
<synopsis>Persists SIP subscriptions so they survive restarts.</synopsis>
<configOption name="packet">
<synopsis>Entire SIP SUBSCRIBE packet that created the subscription</synopsis>
</configOption>
<configOption name="src_name">
<synopsis>The source address of the subscription</synopsis>
</configOption>
<configOption name="src_port">
<synopsis>The source port of the subscription</synopsis>
</configOption>
<configOption name="transport_key">
<synopsis>The type of transport the subscription was received on</synopsis>
</configOption>
<configOption name="local_name">
<synopsis>The local address the subscription was received on</synopsis>
</configOption>
<configOption name="local_port">
<synopsis>The local port the subscription was received on</synopsis>
</configOption>
<configOption name="cseq">
<synopsis>The sequence number of the next NOTIFY to be sent</synopsis>
</configOption>
<configOption name="tag">
<synopsis>The local tag of the dialog for the subscription</synopsis>
</configOption>
<configOption name="endpoint">
<synopsis>The name of the endpoint that subscribed</synopsis>
</configOption>
<configOption name="expires">
<synopsis>The time at which the subscription expires</synopsis>
</configOption>
<configOption name="contact_uri">
<synopsis>The Contact URI of the dialog for the subscription</synopsis>
</configOption>
</configObject>
<configObject name="resource_list">
<synopsis>Resource list configuration parameters.</synopsis>
<description>
<para>This configuration object allows for RFC 4662 resource list subscriptions
to be specified. This can be useful to decrease the amount of subscription traffic
that a server has to process.</para>
<note>
<para>Current limitations limit the size of SIP NOTIFY requests that Asterisk sends
to 64000 bytes. If your resource list notifications are larger than this maximum, you
will need to make adjustments.</para>
</note>
</description>
<configOption name="type">
<synopsis>Must be of type 'resource_list'</synopsis>
</configOption>
<configOption name="event">
<synopsis>The SIP event package that the list resource belong to.</synopsis>
<description>
<para>
The SIP event package describes the types of resources that Asterisk reports
the state of.
</para>
<enumlist>
<enum name="presence">
<para>
Device state and presence reporting.
</para>
</enum>
<enum name="dialog">
<para>
This is identical to <replaceable>presence</replaceable>.
</para>
</enum>
<enum name="message-summary">
<para>
Message-waiting indication (MWI) reporting.
</para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="list_item">
<synopsis>The name of a resource to report state on</synopsis>
<description>
<para>In general Asterisk looks up list items in the following way:</para>
<para>1. Check if the list item refers to another configured resource list.</para>
<para>2. Pass the name of the resource off to event-package-specific handlers
to find the specified resource.</para>
<para>The second part means that the way the list item is specified depends
on what type of list this is. For instance, if you have the <replaceable>event</replaceable>
set to <literal>presence</literal>, then list items should be in the form of
dialplan_extension@dialplan_context. For <literal>message-summary</literal> mailbox
names should be listed.</para>
</description>
</configOption>
<configOption name="full_state" default="no">
<synopsis>Indicates if the entire list's state should be sent out.</synopsis>
<description>
<para>If this option is enabled, and a resource changes state, then Asterisk will construct
a notification that contains the state of all resources in the list. If the option is
disabled, Asterisk will construct a notification that only contains the states of
resources that have changed.</para>
<note>
<para>Even with this option disabled, there are certain situations where Asterisk is forced
to send a notification with the states of all resources in the list. When a subscriber
renews or terminates its subscription to the list, Asterisk MUST send a full state
notification.</para>
</note>
</description>
</configOption>
<configOption name="notification_batch_interval" default="0">
<synopsis>Time Asterisk should wait, in milliseconds, before sending notifications.</synopsis>
<description>
<para>When a resource's state changes, it may be desired to wait a certain amount before Asterisk
sends a notification to subscribers. This allows for other state changes to accumulate, so that
Asterisk can communicate multiple state changes in a single notification instead of rapidly sending
many notifications.</para>
</description>
</configOption>
</configObject>
<configObject name="inbound-publication">
<synopsis>The configuration for inbound publications</synopsis>
<configOption name="endpoint" default="">
<synopsis>Optional name of an endpoint that is only allowed to publish to this resource</synopsis>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'inbound-publication'.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<application name="Park" language="en_US">
<synopsis>
Park yourself.
</synopsis>
<syntax>
<parameter name="parking_lot_name">
<para>Specify in which parking lot to park a call.</para>
<para>The parking lot used is selected in the following order:</para>
<para>1) parking_lot_name option to this application</para>
<para>2) <variable>PARKINGLOT</variable> variable</para>
<para>3) <literal>CHANNEL(parkinglot)</literal> function
(Possibly preset by the channel driver.)</para>
<para>4) Default parking lot.</para>
</parameter>
<parameter name="options">
<para>A list of options for this parked call.</para>
<optionlist>
<option name="r">
<para>Send ringing instead of MOH to the parked call.</para>
</option>
<option name="R">
<para>Randomize the selection of a parking space.</para>
</option>
<option name="s">
<para>Silence announcement of the parking space number.</para>
</option>
<option name="c" argsep=",">
<argument name="context" required="false" />
<argument name="extension" required="false" />
<argument name="priority" required="true" />
<para>If the parking times out, go to this place in the dialplan
instead of where the parking lot defines the call should go.
</para>
</option>
<option name="t">
<argument name="duration" required="true" />
<para>Use a timeout of <literal>duration</literal> seconds instead
of the timeout specified by the parking lot.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Used to park yourself (typically in combination with an attended
transfer to know the parking space).</para>
<para>If you set the <variable>PARKINGEXTEN</variable> variable to a
parking space extension in the parking lot, Park() will attempt to park the
call on that extension. If the extension is already in use then execution
will continue at the next priority.
</para>
<para>If the <literal>parkeddynamic</literal> option is enabled in
<filename>res_parking.conf</filename> the following variables can be
used to dynamically create new parking lots. When using dynamic parking
lots, be aware of the conditions as explained in the notes section
below.
</para>
<para>The <variable>PARKINGDYNAMIC</variable> variable specifies the
parking lot to use as a template to create a dynamic parking lot. It
is an error to specify a non-existent parking lot for the template.
If not set then the default parking lot is used as the template.
</para>
<para>The <variable>PARKINGDYNCONTEXT</variable> variable specifies the
dialplan context to use for the newly created dynamic parking lot. If
not set then the context from the parking lot template is used. The
context is created if it does not already exist and the new parking lot
needs to create extensions.
</para>
<para>The <variable>PARKINGDYNEXTEN</variable> variable specifies the
<literal>parkext</literal> to use for the newly created dynamic
parking lot. If not set then the <literal>parkext</literal> is used from
the parking lot template. If the template does not specify a
<literal>parkext</literal> then no extensions are created for the newly
created parking lot. The dynamic parking lot cannot be created if it
needs to create extensions that overlap existing parking lot extensions.
The only exception to this is for the <literal>parkext</literal>
extension and only if neither of the overlaping parking lot's
<literal>parkext</literal> is exclusive.
</para>
<para>The <variable>PARKINGDYNPOS</variable> variable specifies the
parking positions to use for the newly created dynamic parking lot. If
not set then the <literal>parkpos</literal> from the parking lot template
is used.
</para>
<note>
<para>This application must be used as the first extension priority
to be recognized as a parking access extension for blind transfers.
Blind transfers and the DTMF one-touch parking feature need this
distinction to operate properly. The parking access extension in
this case is treated like a dialplan hint.
</para>
</note>
</description>
<see-also>
<ref type="application">ParkedCall</ref>
</see-also>
</application>
<application name="ParkedCall" language="en_US">
<synopsis>
Retrieve a parked call.
</synopsis>
<syntax>
<parameter name="parking_lot_name">
<para>Specify from which parking lot to retrieve a parked call.</para>
<para>The parking lot used is selected in the following order:</para>
<para>1) parking_lot_name option</para>
<para>2) <variable>PARKINGLOT</variable> variable</para>
<para>3) <literal>CHANNEL(parkinglot)</literal> function
(Possibly preset by the channel driver.)</para>
<para>4) Default parking lot.</para>
</parameter>
<parameter name="parking_space">
<para>Parking space to retrieve a parked call from.
If not provided then the first available parked call in the
parking lot will be retrieved.</para>
</parameter>
</syntax>
<description>
<para>Used to retrieve a parked call from a parking lot.</para>
<note>
<para>If a parking lot's parkext option is set, then Parking lots
will automatically create and manage dialplan extensions in
the parking lot context. If that is the case then you will not
need to manage parking extensions yourself, just include the
parking context of the parking lot.</para>
</note>
</description>
<see-also>
<ref type="application">Park</ref>
</see-also>
</application>
<application name="ParkAndAnnounce" language="en_US">
<synopsis>
Park and Announce.
</synopsis>
<syntax>
<parameter name="parking_lot_name">
<para>Specify in which parking lot to park a call.</para>
<para>The parking lot used is selected in the following order:</para>
<para>1) parking_lot_name option to this application</para>
<para>2) <variable>PARKINGLOT</variable> variable</para>
<para>3) <literal>CHANNEL(parkinglot)</literal> function
(Possibly preset by the channel driver.)</para>
<para>4) Default parking lot.</para>
</parameter>
<parameter name="options">
<para>A list of options for this parked call.</para>
<optionlist>
<option name="r">
<para>Send ringing instead of MOH to the parked call.</para>
</option>
<option name="R">
<para>Randomize the selection of a parking space.</para>
</option>
<option name="c" argsep=",">
<argument name="context" required="false" />
<argument name="extension" required="false" />
<argument name="priority" required="true" />
<para>If the parking times out, go to this place in the dialplan
instead of where the parking lot defines the call should go.
</para>
</option>
<option name="t">
<argument name="duration" required="true" />
<para>Use a timeout of <literal>duration</literal> seconds instead
of the timeout specified by the parking lot.</para>
</option>
</optionlist>
</parameter>
<parameter name="announce_template" required="true" argsep=":">
<argument name="announce" required="true">
<para>Colon-separated list of files to announce. The word
<literal>PARKED</literal> will be replaced by a say_digits of the extension in which
the call is parked.</para>
</argument>
<argument name="announce1" multiple="true" />
</parameter>
<parameter name="dial" required="true">
<para>The app_dial style resource to call to make the
announcement. Console/dsp calls the console.</para>
</parameter>
</syntax>
<description>
<para>Park a call into the parkinglot and announce the call to another channel.</para>
<para>The variable <variable>PARKEDAT</variable> will contain the parking extension
into which the call was placed. Use with the Local channel to allow the dialplan to make
use of this information.</para>
</description>
<see-also>
<ref type="application">Park</ref>
<ref type="application">ParkedCall</ref>
</see-also>
</application>
<manager name="Parkinglots" language="en_US">
<synopsis>
Get a list of parking lots
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>List all parking lots as a series of AMI events</para>
</description>
</manager>
<manager name="ParkedCalls" language="en_US">
<synopsis>
List parked calls.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="ParkingLot">
<para>If specified, only show parked calls from the parking lot with this name.</para>
</parameter>
</syntax>
<description>
<para>List parked calls.</para>
</description>
</manager>
<manager name="Park" language="en_US">
<synopsis>
Park a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to park.</para>
</parameter>
<parameter name="TimeoutChannel" required="false">
<para>Channel name to use when constructing the dial string that will be dialed if the parked channel
times out. If <literal>TimeoutChannel</literal> is in a two party bridge with
<literal>Channel</literal>, then <literal>TimeoutChannel</literal> will receive an announcement and be
treated as having parked <literal>Channel</literal> in the same manner as the Park Call DTMF feature.
</para>
</parameter>
<parameter name="AnnounceChannel" required="false">
<para>If specified, then this channel will receive an announcement when <literal>Channel</literal>
is parked if <literal>AnnounceChannel</literal> is in a state where it can receive announcements
(AnnounceChannel must be bridged). <literal>AnnounceChannel</literal> has no bearing on the actual
state of the parked call.</para>
</parameter>
<parameter name="Timeout" required="false">
<para>Overrides the timeout of the parking lot for this park action. Specified in milliseconds, but will be converted to
seconds. Use a value of 0 to disable the timeout.
</para>
</parameter>
<parameter name="Parkinglot" required="false">
<para>The parking lot to use when parking the channel</para>
</parameter>
</syntax>
<description>
<para>Park an arbitrary channel with optional arguments for specifying the parking lot used, how long
the channel should remain parked, and what dial string to use as the parker if the call times out.
</para>
</description>
</manager>
<managerEvent language="en_US" name="ParkedCall">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel is parked.</synopsis>
<syntax>
<channel_snapshot prefix="Parkee"/>
<parameter name="ParkerDialString">
<para>Dial String that can be used to call back the parker on ParkingTimeout.</para>
</parameter>
<parameter name="Parkinglot">
<para>Name of the parking lot that the parkee is parked in</para>
</parameter>
<parameter name="ParkingSpace">
<para>Parking Space that the parkee is parked in</para>
</parameter>
<parameter name="ParkingTimeout">
<para>Time remaining until the parkee is forcefully removed from parking in seconds</para>
</parameter>
<parameter name="ParkingDuration">
<para>Time the parkee has been in the parking bridge (in seconds)</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ParkedCallTimeOut">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel leaves a parking lot due to reaching the time limit of being parked.</synopsis>
<syntax>
<channel_snapshot prefix="Parkee"/>
<channel_snapshot prefix="Parker"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ParkedCall']/managerEventInstance/syntax/parameter)" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ParkedCallGiveUp">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel leaves a parking lot because it hung up without being answered.</synopsis>
<syntax>
<channel_snapshot prefix="Parkee"/>
<channel_snapshot prefix="Parker"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ParkedCall']/managerEventInstance/syntax/parameter)" />
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="UnParkedCall">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel leaves a parking lot because it was retrieved from the parking lot and reconnected.</synopsis>
<syntax>
<channel_snapshot prefix="Parkee"/>
<channel_snapshot prefix="Parker"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ParkedCall']/managerEventInstance/syntax/parameter)" />
<channel_snapshot prefix="Retriever"/>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="ParkedCallSwap">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a channel takes the place of a previously parked channel</synopsis>
<syntax>
<channel_snapshot prefix="Parkee"/>
<channel_snapshot prefix="Parker"/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ParkedCall']/managerEventInstance/syntax/parameter)" />
</syntax>
<description>
<para>This event is raised when a channel initially parked in the parking lot
is swapped out with a different channel. The most common case for this is when
an attended transfer to a parking lot occurs. The Parkee information in the event
will indicate the party that was swapped into the parking lot.</para>
</description>
</managerEventInstance>
</managerEvent>
<application name="JabberSend" language="en_US" module="res_xmpp">
<synopsis>
Sends an XMPP message to a buddy.
</synopsis>
<syntax>
<parameter name="account" required="true">
<para>The local named account to listen on (specified in
xmpp.conf)</para>
</parameter>
<parameter name="jid" required="true">
<para>Jabber ID of the buddy to send the message to. It can be a
bare JID (username@domain) or a full JID (username@domain/resource).</para>
</parameter>
<parameter name="message" required="true">
<para>The message to send.</para>
</parameter>
</syntax>
<description>
<para>Sends the content of <replaceable>message</replaceable> as text message
from the given <replaceable>account</replaceable> to the buddy identified by
<replaceable>jid</replaceable>
</para>
<para>Example: JabberSend(asterisk,bob@domain.com,Hello world) sends "Hello world"
to <replaceable>bob@domain.com</replaceable> as an XMPP message from the account
<replaceable>asterisk</replaceable>, configured in xmpp.conf.</para>
</description>
<see-also>
<ref type="function" module="res_xmpp">JABBER_STATUS</ref>
<ref type="function" module="res_xmpp">JABBER_RECEIVE</ref>
</see-also>
</application>
<function name="JABBER_RECEIVE" language="en_US" module="res_xmpp">
<synopsis>
Reads XMPP messages.
</synopsis>
<syntax>
<parameter name="account" required="true">
<para>The local named account to listen on (specified in
xmpp.conf)</para>
</parameter>
<parameter name="jid" required="true">
<para>Jabber ID of the buddy to receive message from. It can be a
bare JID (username@domain) or a full JID (username@domain/resource).</para>
</parameter>
<parameter name="timeout">
<para>In seconds, defaults to <literal>20</literal>.</para>
</parameter>
</syntax>
<description>
<para>Receives a text message on the given <replaceable>account</replaceable>
from the buddy identified by <replaceable>jid</replaceable> and returns the contents.</para>
<para>Example: ${JABBER_RECEIVE(asterisk,bob@domain.com)} returns an XMPP message
sent from <replaceable>bob@domain.com</replaceable> (or nothing in case of a time out), to
the <replaceable>asterisk</replaceable> XMPP account configured in xmpp.conf.</para>
</description>
<see-also>
<ref type="function" module="res_xmpp">JABBER_STATUS</ref>
<ref type="application" module="res_xmpp">JabberSend</ref>
</see-also>
</function>
<function name="JABBER_STATUS" language="en_US" module="res_xmpp">
<synopsis>
Retrieves a buddy's status.
</synopsis>
<syntax>
<parameter name="account" required="true">
<para>The local named account to listen on (specified in
xmpp.conf)</para>
</parameter>
<parameter name="jid" required="true">
<para>Jabber ID of the buddy to receive message from. It can be a
bare JID (username@domain) or a full JID (username@domain/resource).</para>
</parameter>
</syntax>
<description>
<para>Retrieves the numeric status associated with the buddy identified
by <replaceable>jid</replaceable>.
If the buddy does not exist in the buddylist, returns 7.</para>
<para>Status will be 1-7.</para>
<para>1=Online, 2=Chatty, 3=Away, 4=XAway, 5=DND, 6=Offline</para>
<para>If not in roster variable will be set to 7.</para>
<para>Example: ${JABBER_STATUS(asterisk,bob@domain.com)} returns 1 if
<replaceable>bob@domain.com</replaceable> is online. <replaceable>asterisk</replaceable> is
the associated XMPP account configured in xmpp.conf.</para>
</description>
<see-also>
<ref type="function" module="res_xmpp">JABBER_RECEIVE</ref>
<ref type="application" module="res_xmpp">JabberSend</ref>
</see-also>
</function>
<application name="JabberSendGroup" language="en_US" module="res_xmpp">
<synopsis>
Send a Jabber Message to a specified chat room
</synopsis>
<syntax>
<parameter name="Jabber" required="true">
<para>Client or transport Asterisk uses to connect to Jabber.</para>
</parameter>
<parameter name="RoomJID" required="true">
<para>XMPP/Jabber JID (Name) of chat room.</para>
</parameter>
<parameter name="Message" required="true">
<para>Message to be sent to the chat room.</para>
</parameter>
<parameter name="Nickname" required="false">
<para>The nickname Asterisk uses in the chat room.</para>
</parameter>
</syntax>
<description>
<para>Allows user to send a message to a chat room via XMPP.</para>
<note>
<para>To be able to send messages to a chat room, a user must have previously joined it. Use the <replaceable>JabberJoin</replaceable> function to do so.</para>
</note>
</description>
</application>
<application name="JabberJoin" language="en_US" module="res_xmpp">
<synopsis>
Join a chat room
</synopsis>
<syntax>
<parameter name="Jabber" required="true">
<para>Client or transport Asterisk uses to connect to Jabber.</para>
</parameter>
<parameter name="RoomJID" required="true">
<para>XMPP/Jabber JID (Name) of chat room.</para>
</parameter>
<parameter name="Nickname" required="false">
<para>The nickname Asterisk will use in the chat room.</para>
<note>
<para>If a different nickname is supplied to an already joined room, the old nick will be changed to the new one.</para>
</note>
</parameter>
</syntax>
<description>
<para>Allows Asterisk to join a chat room.</para>
</description>
</application>
<application name="JabberLeave" language="en_US" module="res_xmpp">
<synopsis>
Leave a chat room
</synopsis>
<syntax>
<parameter name="Jabber" required="true">
<para>Client or transport Asterisk uses to connect to Jabber.</para>
</parameter>
<parameter name="RoomJID" required="true">
<para>XMPP/Jabber JID (Name) of chat room.</para>
</parameter>
<parameter name="Nickname" required="false">
<para>The nickname Asterisk uses in the chat room.</para>
</parameter>
</syntax>
<description>
<para>Allows Asterisk to leave a chat room.</para>
</description>
</application>
<application name="JabberStatus" language="en_US" module="res_xmpp">
<synopsis>
Retrieve the status of a jabber list member
</synopsis>
<syntax>
<parameter name="Jabber" required="true">
<para>Client or transport Asterisk users to connect to Jabber.</para>
</parameter>
<parameter name="JID" required="true">
<para>XMPP/Jabber JID (Name) of recipient.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Variable to store the status of requested user.</para>
</parameter>
</syntax>
<description>
<para>This application is deprecated. Please use the JABBER_STATUS() function instead.</para>
<para>Retrieves the numeric status associated with the specified buddy <replaceable>JID</replaceable>.
The return value in the <replaceable>Variable</replaceable>will be one of the following.</para>
<enumlist>
<enum name="1">
<para>Online.</para>
</enum>
<enum name="2">
<para>Chatty.</para>
</enum>
<enum name="3">
<para>Away.</para>
</enum>
<enum name="4">
<para>Extended Away.</para>
</enum>
<enum name="5">
<para>Do Not Disturb.</para>
</enum>
<enum name="6">
<para>Offline.</para>
</enum>
<enum name="7">
<para>Not In Roster.</para>
</enum>
</enumlist>
</description>
</application>
<manager name="JabberSend" language="en_US" module="res_xmpp">
<synopsis>
Sends a message to a Jabber Client.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Jabber" required="true">
<para>Client or transport Asterisk uses to connect to JABBER.</para>
</parameter>
<parameter name="JID" required="true">
<para>XMPP/Jabber JID (Name) of recipient.</para>
</parameter>
<parameter name="Message" required="true">
<para>Message to be sent to the buddy.</para>
</parameter>
</syntax>
<description>
<para>Sends a message to a Jabber Client.</para>
</description>
</manager>
<info name="MessageToInfo" language="en_US" tech="XMPP">
<para>Specifying a prefix of <literal>xmpp:</literal> will send the
message as an XMPP chat message.</para>
</info>
<info name="MessageFromInfo" language="en_US" tech="XMPP">
<para>Specifying a prefix of <literal>xmpp:</literal> will specify the
account defined in <literal>xmpp.conf</literal> to send the message from.
Note that this field is required for XMPP messages.</para>
</info>
<configInfo name="res_xmpp" language="en_US">
<synopsis>XMPP Messaging</synopsis>
<configFile name="xmpp.conf">
<configObject name="global">
<synopsis>Global configuration settings</synopsis>
<configOption name="debug">
<synopsis>Enable/disable XMPP message debugging</synopsis>
</configOption>
<configOption name="autoprune">
<synopsis>Auto-remove users from buddy list.</synopsis>
<description>
<para>Auto-remove users from buddy list. Depending on the setup
(e.g., using your personal Gtalk account for a test) this could cause loss of
the contact list.
</para>
</description>
</configOption>
<configOption name="autoregister">
<synopsis>Auto-register users from buddy list</synopsis>
</configOption>
<configOption name="collection_nodes">
<synopsis>Enable support for XEP-0248 for use with distributed device state</synopsis>
</configOption>
<configOption name="pubsub_autocreate">
<synopsis>Whether or not the PubSub server supports/is using auto-create for nodes</synopsis>
</configOption>
<configOption name="auth_policy">
<synopsis>Whether to automatically accept or deny users' subscription requests</synopsis>
</configOption>
</configObject>
<configObject name="client">
<synopsis>Configuration options for an XMPP client</synopsis>
<configOption name="username">
<synopsis>XMPP username with optional resource</synopsis>
</configOption>
<configOption name="secret">
<synopsis>XMPP password</synopsis>
</configOption>
<configOption name="refresh_token">
<synopsis>Google OAuth 2.0 refresh token</synopsis>
</configOption>
<configOption name="oauth_clientid">
<synopsis>Google OAuth 2.0 application's client id</synopsis>
</configOption>
<configOption name="oauth_secret">
<synopsis>Google OAuth 2.0 application's secret</synopsis>
</configOption>
<configOption name="serverhost">
<synopsis>Route to server, e.g. talk.google.com</synopsis>
</configOption>
<configOption name="statusmessage">
<synopsis>Custom status message</synopsis>
</configOption>
<configOption name="pubsub_node">
<synopsis>Node for publishing events via PubSub</synopsis>
</configOption>
<configOption name="context">
<synopsis>Dialplan context to send incoming messages to</synopsis>
</configOption>
<configOption name="priority">
<synopsis>XMPP resource priority</synopsis>
</configOption>
<configOption name="port">
<synopsis>XMPP server port</synopsis>
</configOption>
<configOption name="timeout">
<synopsis>Timeout in seconds to hold incoming messages</synopsis>
<description>
<para>Timeout (in seconds) on the message stack. Messages stored longer
than this value will be deleted by Asterisk. This option applies to incoming messages only
which are intended to be processed by the <literal>JABBER_RECEIVE</literal> dialplan function.
</para>
</description>
</configOption>
<configOption name="debug">
<synopsis>Enable debugging</synopsis>
</configOption>
<configOption name="type">
<synopsis>Connection is either a client or a component</synopsis>
</configOption>
<configOption name="distribute_events">
<synopsis>Whether or not to distribute events using this connection</synopsis>
</configOption>
<configOption name="usetls">
<synopsis>Whether to use TLS for the connection or not</synopsis>
</configOption>
<configOption name="usesasl">
<synopsis>Whether to use SASL for the connection or not</synopsis>
</configOption>
<configOption name="forceoldssl">
<synopsis>Force the use of old-style SSL for the connection</synopsis>
</configOption>
<configOption name="keepalive">
<synopsis>If enabled, periodically send an XMPP message from this client with an empty message</synopsis>
</configOption>
<configOption name="autoprune">
<synopsis>Auto-remove users from buddy list.</synopsis>
<description>
<para>Auto-remove users from buddy list. Depending on the setup
(e.g., using your personal Gtalk account for a test) this could cause loss of
the contact list.
</para>
</description>
</configOption>
<configOption name="autoregister">
<synopsis>Auto-register users bfrom buddy list</synopsis>
</configOption>
<configOption name="auth_policy">
<synopsis>Whether to automatically accept or deny users' subscription requests</synopsis>
</configOption>
<configOption name="sendtodialplan">
<synopsis>Send incoming messages into the dialplan</synopsis>
</configOption>
<configOption name="status">
<synopsis>Default XMPP status for the client</synopsis>
<description>
<para>Can be one of the following XMPP statuses:</para>
<enumlist>
<enum name="chat"/>
<enum name="available"/>
<enum name="away"/>
<enum name="xaway"/>
<enum name="dnd"/>
</enumlist>
</description>
</configOption>
<configOption name="buddy">
<synopsis>Manual addition of buddy to list</synopsis>
<description>
<para>
Manual addition of buddy to the buddy list. For distributed events, these budies are
automatically added in the whitelist as 'owners' of the node(s).
</para>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_pjsip_phoneprov_provider" language="en_US">
<synopsis>Module that integrates res_pjsip with res_phoneprov.</synopsis>
<description>
<para>
<emphasis>PJSIP Phoneprov Provider</emphasis>
</para>
<para>This module creates the integration between <literal>res_pjsip</literal> and
<literal>res_phoneprov</literal>.
</para>
<para>Each user to be integrated requires a <literal>phoneprov</literal>
section defined in <filename>pjsip.conf</filename>. Each section identifies
the endpoint associated with the user and any other name/value pairs to be passed
on to res_phoneprov's template substitution. Only <literal>MAC</literal> and
<literal>PROFILE</literal> variables are required. Any other variables
supplied will be passed through.</para>
<para> </para>
<para>Example:</para>
<para>[1000]</para>
<para>type = phoneprovr</para>
<para>endpoint = ep1000</para>
<para>MAC = deadbeef4dad</para>
<para>PROFILE = grandstream2</para>
<para>LINEKEYS = 2</para>
<para>LINE = 1</para>
<para>OTHERVAR = othervalue</para>
<para> </para>
<para>The following variables are automatically defined if an endpoint
is defined for the user:</para>
<enumlist>
<enum name="USERNAME">
<para>Source: The user_name defined in the first auth reference
in the endpoint.</para>
</enum>
<enum name="SECRET">
<para>Source: The user_pass defined in the first auth reference
in the endpoint.</para>
</enum>
<enum name="CALLERID">
<para>Source: The number part of the callerid defined in
the endpoint.</para>
</enum>
<enum name="DISPLAY_NAME">
<para>Source: The name part of the callerid defined in
the endpoint.</para>
</enum>
<enum name="LABEL">
<para>Source: The id of the phoneprov section.</para>
</enum>
</enumlist>
<para> </para>
<para>In addition to the standard variables, the following are also automatically defined:</para>
<enumlist>
<enum name="ENDPOINT_ID">
<para>Source: The id of the endpoint.</para>
</enum>
<enum name="TRANSPORT_ID">
<para>Source: The id of the transport used by the endpoint.</para>
</enum>
<enum name="AUTH_ID">
<para>Source: The id of the auth used by the endpoint.</para>
</enum>
</enumlist>
<para> </para>
<para>All other template substitution variables must be explicitly defined in the
phoneprov_default or phoneprov sections.</para>
</description>
<configFile name="pjsip.conf">
<configObject name="phoneprov">
<synopsis>Provides variables for each user.</synopsis>
<configOption name="type">
<synopsis>Must be of type 'phoneprov'.</synopsis>
</configOption>
<configOption name="endpoint">
<synopsis>The endpoint from which variables will be retrieved.</synopsis>
</configOption>
<configOption name="MAC">
<synopsis>The mac address for this user. (required)</synopsis>
</configOption>
<configOption name="PROFILE">
<synopsis>The phoneprov profile to use for this user. (required)</synopsis>
</configOption>
<configOption name="*">
<synopsis>Other name/value pairs to be passed through for use in templates.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="PresenceStateList" language="en_US">
<synopsis>
List the current known presence states.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>This will list out all known presence states in a
sequence of <replaceable>PresenceStateChange</replaceable> events.
When finished, a <replaceable>PresenceStateListComplete</replaceable> event
will be emitted.</para>
</description>
<see-also>
<ref type="manager">PresenceState</ref>
<ref type="managerEvent">PresenceStatus</ref>
<ref type="function">PRESENCE_STATE</ref>
</see-also>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='PresenceStateChange'])" />
</list-elements>
<managerEvent name="PresenceStateListComplete" language="en_US">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>
Indicates the end of the list the current known extension states.
</synopsis>
<syntax>
<parameter name="EventList">
<para>Conveys the status of the event list.</para>
</parameter>
<parameter name="ListItems">
<para>Conveys the number of statuses reported.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<function name="PP_EACH_EXTENSION" language="en_US">
<synopsis>
Execute specified template for each extension.
</synopsis>
<syntax>
<parameter name="mac" required="true" />
<parameter name="template_file" required="true" />
</syntax>
<description>
<para>Output the specified template for each extension associated with the specified MAC address.</para>
</description>
</function>
<function name="PP_EACH_USER" language="en_US">
<synopsis>
Generate a string for each phoneprov user.
</synopsis>
<syntax>
<parameter name="string" required="true" />
<parameter name="exclude_mac" required="true" />
</syntax>
<description>
<para>Pass in a string, with phoneprov variables you want substituted in the format of
%{VARNAME}, and you will get the string rendered for each user in phoneprov
excluding ones with MAC address <replaceable>exclude_mac</replaceable>. Probably not
useful outside of res_phoneprov.</para>
<para>Example: ${PP_EACH_USER(&lt;item&gt;&lt;fn&gt;%{DISPLAY_NAME}&lt;/fn&gt;&lt;/item&gt;|${MAC})</para>
</description>
</function>
<function name="PJSIP_HEADER" language="en_US">
<synopsis>
Gets headers from an inbound PJSIP channel. Adds, updates or removes the
specified SIP header from an outbound PJSIP channel.
</synopsis>
<syntax>
<parameter name="action" required="true">
<enumlist>
<enum name="read">
<para>Returns instance <replaceable>number</replaceable>
of header <replaceable>name</replaceable>.</para>
</enum>
<enum name="add">
<para>Adds a new header <replaceable>name</replaceable>
to this session.</para>
</enum>
<enum name="update">
<para>Updates instance <replaceable>number</replaceable>
of header <replaceable>name</replaceable> to a new value.
The header must already exist.</para>
</enum>
<enum name="remove">
<para>Removes all instances of previously added headers
whose names match <replaceable>name</replaceable>. A <literal>*</literal>
may be appended to <replaceable>name</replaceable> to remove all headers
<emphasis>beginning with</emphasis>
<replaceable>name</replaceable>.
<replaceable>name</replaceable> may be set to a single <literal>*</literal>
to clear <emphasis>all</emphasis> previously added headers. In all cases,
the number of headers actually removed is returned.</para>
</enum>
</enumlist>
</parameter>
<parameter name="name" required="true">
<para>The name of the header.</para>
</parameter>
<parameter name="number" required="false">
<para>If there's more than 1 header with the same name, this specifies which header
to read or update. If not specified, defaults to <literal>1</literal> meaning
the first matching header. Not valid for <literal>add</literal> or
<literal>remove</literal>.</para>
</parameter>
</syntax>
<description>
<para>PJSIP_HEADER allows you to read specific SIP headers from the inbound
PJSIP channel as well as write(add, update, remove) headers on the outbound
channel. One exception is that you can read headers that you have already
added on the outbound channel.</para>
<para>Examples:</para>
<para>;</para>
<para>; Set 'somevar' to the value of the 'From' header.</para>
<para>exten => 1,1,Set(somevar=${PJSIP_HEADER(read,From)})</para>
<para>;</para>
<para>; Set 'via2' to the value of the 2nd 'Via' header.</para>
<para>exten => 1,1,Set(via2=${PJSIP_HEADER(read,Via,2)})</para>
<para>;</para>
<para>; Add an 'X-Myheader' header with the value of 'myvalue'.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(add,X-MyHeader)=myvalue)</para>
<para>;</para>
<para>; Add an 'X-Myheader' header with an empty value.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(add,X-MyHeader)=)</para>
<para>;</para>
<para>; Update the value of the header named 'X-Myheader' to 'newvalue'.</para>
<para>; 'X-Myheader' must already exist or the call will fail.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(update,X-MyHeader)=newvalue)</para>
<para>;</para>
<para>; Remove all headers whose names exactly match 'X-MyHeader'.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(remove,X-MyHeader)=)</para>
<para>;</para>
<para>; Remove all headers that begin with 'X-My'.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(remove,X-My*)=)</para>
<para>;</para>
<para>; Remove all previously added headers.</para>
<para>exten => 1,1,Set(PJSIP_HEADER(remove,*)=)</para>
<para>;</para>
<note>
<para>The <literal>remove</literal> action can be called by reading
<emphasis>or</emphasis> writing PJSIP_HEADER.</para>
<para>;</para>
<para>; Display the number of headers removed</para>
<para>exten => 1,1,Verbose( Removed ${PJSIP_HEADER(remove,X-MyHeader)} headers)</para>
<para>;</para>
<para>; Set a variable to the number of headers removed</para>
<para>exten => 1,1,Set(count=${PJSIP_HEADER(remove,X-MyHeader)})</para>
<para>;</para>
<para>; Just remove them ignoring any count</para>
<para>exten => 1,1,Set(=${PJSIP_HEADER(remove,X-MyHeader)})</para>
<para>exten => 1,1,Set(PJSIP_HEADER(remove,X-MyHeader)=)</para>
<para>;</para>
</note>
<note>
<para>If you call PJSIP_HEADER in a normal dialplan context you'll be
operating on the <emphasis>caller's (incoming)</emphasis> channel which
may not be what you want. To operate on the <emphasis>callee's (outgoing)</emphasis>
channel call PJSIP_HEADER in a pre-dial handler. </para>
<para>Example:</para>
<para>;</para>
<para>[handler]</para>
<para>exten => addheader,1,Set(PJSIP_HEADER(add,X-MyHeader)=myvalue)</para>
<para>exten => addheader,2,Set(PJSIP_HEADER(add,X-MyHeader2)=myvalue2)</para>
<para>;</para>
<para>[somecontext]</para>
<para>exten => 1,1,Dial(PJSIP/${EXTEN},,b(handler^addheader^1))</para>
<para>;</para>
</note>
</description>
</function>
<configInfo name="res_pjproject" language="en_US">
<synopsis>pjproject common configuration</synopsis>
<configFile name="pjproject.conf">
<configObject name="startup">
<synopsis>Asterisk startup time options for PJPROJECT</synopsis>
<description>
<note>
<para>The id of this object, as well as its type, must be
'startup' or it won't be found.</para>
</note>
</description>
<configOption name="type">
<synopsis>Must be of type 'startup'.</synopsis>
</configOption>
<configOption name="log_level" default="2">
<synopsis>Initial maximum pjproject logging level to log.</synopsis>
<description>
<para>Valid values are: 0-6, and default</para>
<note>
<para>
This option is needed very early in the startup process
so it can only be read from config files because the
modules for other methods have not been loaded yet.
</para>
</note>
</description>
</configOption>
</configObject>
<configObject name="log_mappings">
<synopsis>PJPROJECT to Asterisk Log Level Mapping</synopsis>
<description>
<para>Warnings and errors in the pjproject libraries are generally handled
by Asterisk. In many cases, Asterisk wouldn't even consider them to
be warnings or errors so the messages emitted by pjproject directly
are either superfluous or misleading. The 'log_mappings'
object allows mapping the pjproject levels to Asterisk levels, or nothing.
</para>
<note>
<para>The id of this object, as well as its type, must be
'log_mappings' or it won't be found.</para>
</note>
</description>
<configOption name="type">
<synopsis>Must be of type 'log_mappings'.</synopsis>
</configOption>
<configOption name="asterisk_error" default="0,1">
<synopsis>A comma separated list of pjproject log levels to map to Asterisk LOG_ERROR.</synopsis>
</configOption>
<configOption name="asterisk_warning" default="2">
<synopsis>A comma separated list of pjproject log levels to map to Asterisk LOG_WARNING.</synopsis>
</configOption>
<configOption name="asterisk_notice" default="">
<synopsis>A comma separated list of pjproject log levels to map to Asterisk LOG_NOTICE.</synopsis>
</configOption>
<configOption name="asterisk_debug" default="3,4,5,6">
<synopsis>A comma separated list of pjproject log levels to map to Asterisk LOG_DEBUG.</synopsis>
</configOption>
<configOption name="asterisk_verbose" default="">
<synopsis>A comma separated list of pjproject log levels to map to Asterisk LOG_VERBOSE.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<info name="MessageFromInfo" language="en_US" tech="PJSIP">
<para>The <literal>from</literal> parameter can be a configured endpoint
or in the form of "display-name" &lt;URI&gt;.</para>
</info>
<info name="MessageToInfo" language="en_US" tech="PJSIP">
<para>Specifying a prefix of <literal>pjsip:</literal> will send the
message as a SIP MESSAGE request.</para>
</info>
<function name="MUTEAUDIO" language="en_US">
<synopsis>
Muting audio streams in the channel
</synopsis>
<syntax>
<parameter name="direction" required="true">
<para>Must be one of </para>
<enumlist>
<enum name="in">
<para>Inbound stream (to the PBX)</para>
</enum>
<enum name="out">
<para>Outbound stream (from the PBX)</para>
</enum>
<enum name="all">
<para>Both streams</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.
</para>
<para>Examples:
</para>
<para>
MUTEAUDIO(in)=on
</para>
<para>
MUTEAUDIO(in)=off
</para>
</description>
</function>
<manager name="MuteAudio" language="en_US">
<synopsis>
Mute an audio stream.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The channel you want to mute.</para>
</parameter>
<parameter name="Direction" required="true">
<enumlist>
<enum name="in">
<para>Set muting on inbound audio stream. (to the PBX)</para>
</enum>
<enum name="out">
<para>Set muting on outbound audio stream. (from the PBX)</para>
</enum>
<enum name="all">
<para>Set muting on inbound and outbound audio streams.</para>
</enum>
</enumlist>
</parameter>
<parameter name="State" required="true">
<enumlist>
<enum name="on">
<para>Turn muting on.</para>
</enum>
<enum name="off">
<para>Turn muting off.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Mute an incoming or outgoing audio stream on a channel.</para>
</description>
</manager>
<configInfo name="res_parking" language="en_US">
<configFile name="res_parking.conf">
<configObject name="globals">
<synopsis>Options that apply to every parking lot</synopsis>
<configOption name="parkeddynamic">
<synopsis>Enables dynamically created parkinglots.</synopsis>
<description>
<para>If the option is enabled then the following variables can
be used to dynamically create new parking lots.
</para>
<para>The <variable>PARKINGDYNAMIC</variable> variable specifies the
parking lot to use as a template to create a dynamic parking lot. It
is an error to specify a non-existent parking lot for the template.
If not set then the default parking lot is used as the template.
</para>
<para>The <variable>PARKINGDYNCONTEXT</variable> variable specifies the
dialplan context to use for the newly created dynamic parking lot. If
not set then the context from the parking lot template is used. The
context is created if it does not already exist and the new parking lot
needs to create extensions.
</para>
<para>The <variable>PARKINGDYNEXTEN</variable> variable specifies the
<literal>parkext</literal> to use for the newly created dynamic
parking lot. If not set then the <literal>parkext</literal> is used from
the parking lot template. If the template does not specify a
<literal>parkext</literal> then no extensions are created for the newly
created parking lot. The dynamic parking lot cannot be created if it
needs to create extensions that overlap existing parking lot extensions.
The only exception to this is for the <literal>parkext</literal>
extension and only if neither of the overlaping parking lot's
<literal>parkext</literal> is exclusive.
</para>
<para>The <variable>PARKINGDYNPOS</variable> variable specifies the
parking positions to use for the newly created dynamic parking lot. If
not set then the <literal>parkpos</literal> from the parking lot template
is used.
</para>
</description>
</configOption>
</configObject>
<configObject name="parking_lot">
<synopsis>Defined parking lots for res_parking to use to park calls on</synopsis>
<configOption name="context" default="parkedcalls">
<synopsis>The name of the context where calls are parked and picked up from.</synopsis>
<description>
<para>This option is only used if parkext is set.</para>
</description>
</configOption>
<configOption name="parkext">
<synopsis>Extension to park calls to this parking lot.</synopsis>
<description>
<para>If this option is used, this extension will automatically
be created to place calls into parking lots. In addition, if
<literal>parkext_exclusive</literal> is set for this parking
lot, the name of the parking lot will be included in the
application's arguments so that it only parks to this parking
lot. The extension will be created in <literal>context</literal>.
Using this option also creates extensions for retrieving
parked calls from the parking spaces in the same context.
</para>
<note>
<para>Generated parking extensions cannot overlap.
The only exception is if neither overlapping
<literal>parkext</literal> is exclusive.
</para>
</note>
</description>
</configOption>
<configOption name="parkext_exclusive" default="no">
<synopsis>If yes, the extension registered as parkext will park exclusively to this parking lot.</synopsis>
</configOption>
<configOption name="parkpos" default="701-750">
<synopsis>Numerical range of parking spaces which can be used to retrieve parked calls.</synopsis>
<description>
<para>If <literal>parkext</literal> is set, these extensions
will automatically be mapped in <literal>context</literal>
in order to pick up calls parked to these parking spaces.
</para>
</description>
</configOption>
<configOption name="parkinghints" default="no">
<synopsis>If yes, this parking lot will add hints automatically for parking spaces.</synopsis>
</configOption>
<configOption name="parkingtime" default="45">
<synopsis>Amount of time a call will remain parked before giving up (in seconds).</synopsis>
</configOption>
<configOption name="parkedmusicclass">
<synopsis>Which music class to use for parked calls. They will use the default if unspecified.</synopsis>
</configOption>
<configOption name="comebacktoorigin" default="yes">
<synopsis>Determines what should be done with the parked channel if no one picks it up before it times out.</synopsis>
<description>
<para>Valid Options:</para>
<enumlist>
<enum name="yes">
<para>Automatically have the parked channel dial the device that parked the call with dial
timeout set by the <literal>parkingtime</literal> option. When the call times out an extension
to dial the PARKER will automatically be created in the <literal>park-dial</literal> context with
an extension of the flattened parker device name. If the call is not answered, the parked channel
that is timing out will continue in the dial plan at that point if there are more priorities in
the extension (which won't be the case unless the dialplan deliberately includes such priorities
in the <literal>park-dial</literal> context through pattern matching or deliberately written
flattened peer extensions).</para>
</enum>
<enum name="no">
<para>Place the call into the PBX at <literal>comebackcontext</literal> instead. The extension will
still be set as the flattened peer name. If an extension the flattened peer name isn't available
then it will fall back to the <literal>s</literal> extension. If that also is unavailable it will
attempt to fall back to <literal>s@default</literal>. The normal dial extension will still be
created in the <literal>park-dial</literal> context with the extension also being the flattened
peer name.</para>
</enum>
</enumlist>
<note>
<para>Flattened Peer Names - Extensions can not include slash characters since those are used for pattern
matching. When a peer name is flattened, slashes become underscores. For example if the parker of a call
is called <literal>SIP/0004F2040001</literal> then flattened peer name and therefor the extensions created
and used on timeouts will be <literal>SIP_0004F204001</literal>.</para>
</note>
<note>
<para>When parking times out and the channel returns to the dial plan, the following variables are set:
</para>
</note>
<variablelist>
<variable name="PARKING_SPACE">
<para>extension that the call was parked in prior to timing out.</para>
</variable>
<variable name="PARKINGSLOT">
<para>Deprecated. Use <variable>PARKING_SPACE</variable> instead.</para>
</variable>
<variable name="PARKEDLOT">
<para>name of the lot that the call was parked in prior to timing out.</para>
</variable>
<variable name="PARKER">
<para>The device that parked the call</para>
</variable>
<variable name="PARKER_FLAT">
<para>The flat version of <variable>PARKER</variable></para>
</variable>
</variablelist>
</description>
</configOption>
<configOption name="comebackdialtime" default="30">
<synopsis>Timeout for the Dial extension created to call back the parker when a parked call times out.</synopsis>
</configOption>
<configOption name="comebackcontext" default="parkedcallstimeout">
<synopsis>Context where parked calls will enter the PBX on timeout when comebacktoorigin=no</synopsis>
<description>
<para>The extension the call enters will prioritize the flattened peer name in this context.
If the flattened peer name extension is unavailable, then the 's' extension in this context will be
used. If that also is unavailable, the 's' extension in the 'default' context will be used.</para>
</description>
</configOption>
<configOption name="courtesytone">
<synopsis>If the name of a sound file is provided, use this as the courtesy tone</synopsis>
<description>
<para>By default, this tone is only played to the caller of a parked call. Who receives the tone
can be changed using the <literal>parkedplay</literal> option.</para>
</description>
</configOption>
<configOption name="parkedplay" default="caller">
<synopsis>Who we should play the courtesytone to on the pickup of a parked call from this lot</synopsis>
<description>
<enumlist>
<enum name="no">
<para>Apply to neither side.</para>
</enum>
<enum name="caller">
<para>Apply only to the call connecting with the call coming out of the parking lot.</para>
</enum>
<enum name="callee">
<para>Apply only to the call coming out of the parking lot.</para>
</enum>
<enum name="both">
<para>Apply to both sides.</para>
</enum>
</enumlist>
<note>
<para>If courtesy tone is not specified then this option will be ignored.</para>
</note>
</description>
</configOption>
<configOption name="parkedcalltransfers" default="no">
<synopsis>Who to apply the DTMF transfer features to when parked calls are picked up or timeout.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_parking']/configFile[@name='res_parking.conf']/configObject[@name='parking_lot']/configOption[@name='parkedplay']/description/enumlist)" />
</description>
</configOption>
<configOption name="parkedcallreparking" default="no">
<synopsis>Who to apply the DTMF parking feature to when parked calls are picked up or timeout.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_parking']/configFile[@name='res_parking.conf']/configObject[@name='parking_lot']/configOption[@name='parkedplay']/description/enumlist)" />
</description>
</configOption>
<configOption name="parkedcallhangup" default="no">
<synopsis>Who to apply the DTMF hangup feature to when parked calls are picked up or timeout.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_parking']/configFile[@name='res_parking.conf']/configObject[@name='parking_lot']/configOption[@name='parkedplay']/description/enumlist)" />
</description>
</configOption>
<configOption name="parkedcallrecording" default="no">
<synopsis>Who to apply the DTMF MixMonitor recording feature to when parked calls are picked up or timeout.</synopsis>
<description>
<xi:include xpointer="xpointer(/docs/configInfo[@name='res_parking']/configFile[@name='res_parking.conf']/configObject[@name='parking_lot']/configOption[@name='parkedplay']/description/enumlist)" />
</description>
</configOption>
<configOption name="findslot" default="first">
<synopsis>Rule to use when trying to figure out which parking space a call should be parked with.</synopsis>
<description>
<enumlist>
<enum name="first">
<para>Always try to place in the lowest available space in the parking lot</para>
</enum>
<enum name="next">
<para>Track the last parking space used and always attempt to use the one immediately after.
</para>
</enum>
</enumlist>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<application name="Monitor" language="en_US">
<synopsis>
Monitor a channel.
</synopsis>
<syntax>
<parameter name="file_format" argsep=":">
<argument name="file_format" required="true">
<para>Optional. If not set, defaults to <literal>wav</literal></para>
</argument>
<argument name="urlbase" />
</parameter>
<parameter name="fname_base">
<para>If set, changes the filename used to the one specified.</para>
</parameter>
<parameter name="options">
<optionlist>
<option name="m">
<para>When the recording ends mix the two leg files into one and
delete the two leg files. If the variable <variable>MONITOR_EXEC</variable>
is set, the application referenced in it will be executed instead of
soxmix/sox and the raw leg files will NOT be deleted automatically.
soxmix/sox or <variable>MONITOR_EXEC</variable> is handed 3 arguments,
the two leg files and a target mixed file name which is the same as
the leg file names only without the in/out designator.</para>
<para>If <variable>MONITOR_EXEC_ARGS</variable> is set, the contents
will be passed on as additional arguments to <variable>MONITOR_EXEC</variable>.
Both <variable>MONITOR_EXEC</variable> and the Mix flag can be set from the
administrator interface.</para>
<warning>
<para>Do not use untrusted strings such as
<variable>CALLERID(num)</variable> or <variable>CALLERID(name)</variable>
as part of <variable>MONITOR_EXEC</variable> or
<variable>MONITOR_EXEC_ARGS</variable>. You risk a command injection
attack executing arbitrary commands if the untrusted strings aren't
filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para>
</warning>
</option>
<option name="b">
<para>Don't begin recording unless a call is bridged to another channel.</para>
</option>
<option name="B">
<para>Play a periodic beep while this call is being recorded.</para>
<argument name="interval">
<para>Interval, in seconds. Default is 15.</para>
</argument>
</option>
<option name="i">
<para>Skip recording of input stream (disables <literal>m</literal> option).</para>
</option>
<option name="o">
<para>Skip recording of output stream (disables <literal>m</literal> option).</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until the channel hangs up or
monitoring is stopped by the StopMonitor application.</para>
<para>By default, files are stored to <filename>/var/spool/asterisk/monitor/</filename>.
Returns <literal>-1</literal> if monitor files can't be opened or if the channel is
already monitored, otherwise <literal>0</literal>.</para>
</description>
<see-also>
<ref type="application">StopMonitor</ref>
</see-also>
</application>
<application name="StopMonitor" language="en_US">
<synopsis>
Stop monitoring a channel.
</synopsis>
<syntax />
<description>
<para>Stops monitoring a channel. Has no effect if the channel is not monitored.</para>
</description>
</application>
<application name="ChangeMonitor" language="en_US">
<synopsis>
Change monitoring filename of a channel.
</synopsis>
<syntax>
<parameter name="filename_base" required="true">
<para>The new filename base to use for monitoring this channel.</para>
</parameter>
</syntax>
<description>
<para>Changes monitoring filename of a channel. Has no effect if the
channel is not monitored.</para>
</description>
</application>
<application name="PauseMonitor" language="en_US">
<synopsis>
Pause monitoring of a channel.
</synopsis>
<syntax />
<description>
<para>Pauses monitoring of a channel until it is re-enabled by a call to UnpauseMonitor.</para>
</description>
<see-also>
<ref type="application">UnpauseMonitor</ref>
</see-also>
</application>
<application name="UnpauseMonitor" language="en_US">
<synopsis>
Unpause monitoring of a channel.
</synopsis>
<syntax />
<description>
<para>Unpauses monitoring of a channel on which monitoring had
previously been paused with PauseMonitor.</para>
</description>
<see-also>
<ref type="application">PauseMonitor</ref>
</see-also>
</application>
<manager name="Monitor" language="en_US">
<synopsis>
Monitor a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
<parameter name="File">
<para>Is the name of the file created in the monitor spool directory.
Defaults to the same name as the channel (with slashes replaced with dashes).</para>
</parameter>
<parameter name="Format">
<para>Is the audio recording format. Defaults to <literal>wav</literal>.</para>
</parameter>
<parameter name="Mix">
<para>Boolean parameter as to whether to mix the input and output channels
together after the recording is finished.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to record the audio on a
specified channel.</para>
</description>
</manager>
<manager name="StopMonitor" language="en_US">
<synopsis>
Stop monitoring a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel monitored.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to end a previously started 'Monitor' action.</para>
</description>
</manager>
<manager name="ChangeMonitor" language="en_US">
<synopsis>
Change monitoring filename of a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
<parameter name="File" required="true">
<para>Is the new name of the file created in the
monitor spool directory.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to change the file
started by a previous 'Monitor' action.</para>
</description>
</manager>
<manager name="PauseMonitor" language="en_US">
<synopsis>
Pause monitoring of a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to temporarily stop the
recording of a channel.</para>
</description>
</manager>
<manager name="UnpauseMonitor" language="en_US">
<synopsis>
Unpause monitoring of a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to re-enable recording
of a channel after calling PauseMonitor.</para>
</description>
</manager>
<manager name="PJSIPNotify" language="en_US">
<synopsis>
Send a NOTIFY to either an endpoint or an arbitrary URI.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Endpoint" required="false">
<para>The endpoint to which to send the NOTIFY.</para>
</parameter>
<parameter name="URI" required="false">
<para>Abritrary URI to which to send the NOTIFY.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Appends variables as headers/content to the NOTIFY. If the variable is
named <literal>Content</literal>, then the value will compose the body
of the message if another variable sets <literal>Content-Type</literal>.
<replaceable>name</replaceable>=<replaceable>value</replaceable>
</para>
</parameter>
</syntax>
<description>
<para>Sends a NOTIFY to an endpoint or an arbitrary URI.</para>
<para>All parameters for this event must be specified in the body of this
request via multiple <literal>Variable: name=value</literal> sequences.</para>
<note>
<para>One (and only one) of <literal>Endpoint</literal> or
<literal>URI</literal> must be specified. If <literal>URI</literal> is used,
the default outbound endpoint will be used to send the message. If the default
outbound endpoint isn't configured, this command can not send to an arbitrary
URI.</para>
</note>
</description>
</manager>
<configInfo name="res_pjsip_notify" language="en_US">
<synopsis>Module that supports sending NOTIFY requests to endpoints from external sources</synopsis>
<configFile name="pjsip_notify.conf">
<configObject name="general">
<synopsis>Unused, but reserved.</synopsis>
</configObject>
<configObject name="notify">
<synopsis>Configuration of a NOTIFY request.</synopsis>
<description>
<para>Each key-value pair in a <literal>notify</literal>
configuration section defines either a SIP header to send
in the request or a line of content in the request message
body. A key of <literal>Content</literal> is treated
as part of the message body and is appended in sequential
order; any other header is treated as part of the SIP
request.</para>
</description>
<configOption name="^.*$">
<synopsis>A key/value pair to add to a NOTIFY request.</synopsis>
<description>
<para>If the key is <literal>Content</literal>,
it will be treated as part of the message body. Otherwise,
it will be added as a header in the NOTIFY request.</para>
<para>The following headers are reserved and cannot be
specified:</para>
<enumlist>
<enum name="Call-ID" />
<enum name="Contact" />
<enum name="CSeq" />
<enum name="To" />
<enum name="From" />
<enum name="Record-Route" />
<enum name="Route" />
<enum name="Via" />
</enumlist>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<configInfo name="res_ari" language="en_US">
<synopsis>HTTP binding for the Stasis API</synopsis>
<configFile name="ari.conf">
<configObject name="general">
<synopsis>General configuration settings</synopsis>
<configOption name="enabled">
<synopsis>Enable/disable the ARI module</synopsis>
<description>
<para>This option enables or disables the ARI module.</para>
<note>
<para>ARI uses Asterisk's HTTP server, which must also be enabled in <filename>http.conf</filename>.</para>
</note>
</description>
<see-also>
<ref type="filename">http.conf</ref>
<ref type="link">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Builtin+mini-HTTP+Server</ref>
</see-also>
</configOption>
<configOption name="websocket_write_timeout">
<synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
<description>
<para>If a websocket connection accepts input slowly, the timeout
for writes to it can be increased to keep it from being disconnected.
Value is in milliseconds; default is 100 ms.</para>
</description>
</configOption>
<configOption name="pretty">
<synopsis>Responses from ARI are formatted to be human readable</synopsis>
</configOption>
<configOption name="auth_realm">
<synopsis>Realm to use for authentication. Defaults to Asterisk REST Interface.</synopsis>
</configOption>
<configOption name="allowed_origins">
<synopsis>Comma separated list of allowed origins, for Cross-Origin Resource Sharing. May be set to * to allow all origins.</synopsis>
</configOption>
</configObject>
<configObject name="user">
<synopsis>Per-user configuration settings</synopsis>
<configOption name="type">
<synopsis>Define this configuration section as a user.</synopsis>
<description>
<enumlist>
<enum name="user">
<para>Configure this section as a <replaceable>user</replaceable></para>
</enum>
</enumlist>
</description>
</configOption>
<configOption name="read_only">
<synopsis>When set to yes, user is only authorized for read-only requests</synopsis>
</configOption>
<configOption name="password">
<synopsis>Crypted or plaintext password (see password_format)</synopsis>
</configOption>
<configOption name="password_format">
<synopsis>password_format may be set to plain (the default) or crypt. When set to crypt, crypt(3) is used to validate the password. A crypted password can be generated using mkpasswd -m sha-512. When set to plain, the password is in plaintext</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<application name="ReceiveFAX" language="en_US" module="res_fax">
<synopsis>
Receive a FAX and save as a TIFF/F file.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="options">
<optionlist>
<option name="d">
<para>Enable FAX debugging.</para>
</option>
<option name="f">
<para>Allow audio fallback FAX transfer on T.38 capable channels.</para>
</option>
<option name="F">
<para>Force usage of audio mode on T.38 capable channels.</para>
</option>
<option name="s">
<para>Send progress Manager events (overrides statusevents setting in res_fax.conf).</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is provided by res_fax, which is a FAX technology agnostic module
that utilizes FAX technology resource modules to complete a FAX transmission.</para>
<para>Session arguments can be set by the FAXOPT function and to check results of the ReceiveFax() application.</para>
</description>
<see-also>
<ref type="function">FAXOPT</ref>
</see-also>
</application>
<application name="SendFAX" language="en_US" module="res_fax">
<synopsis>
Sends a specified TIFF/F file as a FAX.
</synopsis>
<syntax>
<parameter name="filename" required="true" argsep="&amp;">
<argument name="filename2" multiple="true">
<para>TIFF file to send as a FAX.</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="d">
<para>Enable FAX debugging.</para>
</option>
<option name="f">
<para>Allow audio fallback FAX transfer on T.38 capable channels.</para>
</option>
<option name="F">
<para>Force usage of audio mode on T.38 capable channels.</para>
</option>
<option name="s">
<para>Send progress Manager events (overrides statusevents setting in res_fax.conf).</para>
</option>
<option name="z">
<para>Initiate a T.38 reinvite on the channel if the remote end does not.</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>This application is provided by res_fax, which is a FAX technology agnostic module
that utilizes FAX technology resource modules to complete a FAX transmission.</para>
<para>Session arguments can be set by the FAXOPT function and to check results of the SendFax() application.</para>
</description>
<see-also>
<ref type="function">FAXOPT</ref>
</see-also>
</application>
<function name="FAXOPT" language="en_US" module="res_fax">
<synopsis>
Gets/sets various pieces of information about a fax session.
</synopsis>
<syntax>
<parameter name="item" required="true">
<enumlist>
<enum name="ecm">
<para>R/W Error Correction Mode (ECM) enable with 'yes', disable with 'no'.</para>
</enum>
<enum name="error">
<para>R/O FAX transmission error code upon failure.</para>
</enum>
<enum name="filename">
<para>R/O Filename of the first file of the FAX transmission.</para>
</enum>
<enum name="filenames">
<para>R/O Filenames of all of the files in the FAX transmission (comma separated).</para>
</enum>
<enum name="headerinfo">
<para>R/W FAX header information.</para>
</enum>
<enum name="localstationid">
<para>R/W Local Station Identification.</para>
</enum>
<enum name="minrate">
<para>R/W Minimum transfer rate set before transmission.</para>
</enum>
<enum name="maxrate">
<para>R/W Maximum transfer rate set before transmission.</para>
</enum>
<enum name="modem">
<para>R/W Modem type (v17/v27/v29).</para>
</enum>
<enum name="gateway">
<para>R/W T38 fax gateway, with optional fax activity timeout in seconds (yes[,timeout]/no)</para>
</enum>
<enum name="faxdetect">
<para>R/W Enable FAX detect with optional timeout in seconds (yes,t38,cng[,timeout]/no)</para>
</enum>
<enum name="pages">
<para>R/O Number of pages transferred.</para>
</enum>
<enum name="rate">
<para>R/O Negotiated transmission rate.</para>
</enum>
<enum name="remotestationid">
<para>R/O Remote Station Identification after transmission.</para>
</enum>
<enum name="resolution">
<para>R/O Negotiated image resolution after transmission.</para>
</enum>
<enum name="sessionid">
<para>R/O Session ID of the FAX transmission.</para>
</enum>
<enum name="status">
<para>R/O Result Status of the FAX transmission.</para>
</enum>
<enum name="statusstr">
<para>R/O Verbose Result Status of the FAX transmission.</para>
</enum>
<enum name="t38timeout">
<para>R/W The timeout used for T.38 negotiation.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>FAXOPT can be used to override the settings for a FAX session listed in <filename>res_fax.conf</filename>,
it can also be used to retrieve information about a FAX session that has finished eg. pages/status.</para>
</description>
<see-also>
<ref type="application">ReceiveFax</ref>
<ref type="application">SendFax</ref>
</see-also>
</function>
<manager name="FAXSessions" language="en_US">
<synopsis>
Lists active FAX sessions
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Will generate a series of FAXSession events with information about each FAXSession. Closes with
a FAXSessionsComplete event which includes a count of the included FAX sessions. This action works in
the same manner as the CLI command 'fax show sessions'</para>
</description>
</manager>
<managerEvent language="en_US" name="FAXSessionsEntry">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>A single list item for the FAXSessions AMI command</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="Channel">
<para>Name of the channel responsible for the FAX session</para>
</parameter>
<parameter name="Technology">
<para>The FAX technology that the FAX session is using</para>
</parameter>
<parameter name="SessionNumber">
<para>The numerical identifier for this particular session</para>
</parameter>
<parameter name="SessionType">
<para>FAX session passthru/relay type</para>
<enumlist>
<enum name="G.711" />
<enum name="T.38" />
</enumlist>
</parameter>
<parameter name="Operation">
<para>FAX session operation type</para>
<enumlist>
<enum name="gateway" />
<enum name="V.21" />
<enum name="send" />
<enum name="receive" />
<enum name="none" />
</enumlist>
</parameter>
<parameter name="State">
<para>Current state of the FAX session</para>
<enumlist>
<enum name="Uninitialized" />
<enum name="Initialized" />
<enum name="Open" />
<enum name="Active" />
<enum name="Complete" />
<enum name="Reserved" />
<enum name="Inactive" />
<enum name="Unknown" />
</enumlist>
</parameter>
<parameter name="Files">
<para>File or list of files associated with this FAX session</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="FAXSessionsComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when all FAXSession events are completed for a FAXSessions command</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="Total">
<para>Count of FAXSession events sent in response to FAXSessions action</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="FAXSession" language="en_US">
<synopsis>
Responds with a detailed description of a single FAX session
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="SessionNumber" required="true">
<para>The session ID of the fax the user is interested in.</para>
</parameter>
</syntax>
<description>
<para>Provides details about a specific FAX session. The response will include a common subset of
the output from the CLI command 'fax show session &lt;session_number&gt;' for each technology. If the
FAX technolgy used by this session does not include a handler for FAXSession, then this action
will fail.</para>
</description>
</manager>
<managerEvent language="en_US" name="FAXSession">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>Raised in response to FAXSession manager command</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="SessionNumber">
<para>The numerical identifier for this particular session</para>
</parameter>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FAXSessionsEntry']/managerEventInstance/syntax/parameter[@name='Operation'])" />
<xi:include xpointer="xpointer(/docs/managerEvent[@name='FAXSessionsEntry']/managerEventInstance/syntax/parameter[@name='State'])" />
<parameter name="ErrorCorrectionMode" required="false">
<para>Whether error correcting mode is enabled for the FAX session. This field is not
included when operation is 'V.21 Detect' or if operation is 'gateway' and state is
'Uninitialized'
</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
</enumlist>
</parameter>
<parameter name="DataRate" required="false">
<para>Bit rate of the FAX. This field is not included when operation is 'V.21 Detect' or
if operation is 'gateway' and state is 'Uninitialized'.</para>
</parameter>
<parameter name="ImageResolution" required="false">
<para>Resolution of each page of the FAX. Will be in the format of X_RESxY_RES. This field
is not included if the operation is anything other than Receive/Transmit.</para>
</parameter>
<parameter name="PageNumber" required="false">
<para>Current number of pages transferred during this FAX session. May change as the FAX
progresses. This field is not included when operation is 'V.21 Detect' or if operation is
'gateway' and state is 'Uninitialized'.</para>
</parameter>
<parameter name="FileName" required="false">
<para>Filename of the image being sent/recieved for this FAX session. This field is not
included if Operation isn't 'send' or 'receive'.</para>
</parameter>
<parameter name="PagesTransmitted" required="false">
<para>Total number of pages sent during this session. This field is not included if
Operation isn't 'send' or 'receive'. Will always be 0 for 'receive'.</para>
</parameter>
<parameter name="PagesReceived" required="false">
<para>Total number of pages received during this session. This field is not included if
Operation is not 'send' or 'receive'. Will be 0 for 'send'.</para>
</parameter>
<parameter name="TotalBadLines" required="false">
<para>Total number of bad lines sent/recieved during this session. This field is not
included if Operation is not 'send' or 'received'.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="FAXStats" language="en_US">
<synopsis>
Responds with fax statistics
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Provides FAX statistics including the number of active sessions, reserved sessions, completed
sessions, failed sessions, and the number of receive/transmit attempts. This command provides all
of the non-technology specific information provided by the CLI command 'fax show stats'</para>
</description>
</manager>
<managerEvent language="en_US" name="FAXStats">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>Raised in response to FAXStats manager command</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="CurrentSessions" required="true">
<para>Number of active FAX sessions</para>
</parameter>
<parameter name="ReservedSessions" required="true">
<para>Number of reserved FAX sessions</para>
</parameter>
<parameter name="TransmitAttempts" required="true">
<para>Total FAX sessions for which Asterisk is/was the transmitter</para>
</parameter>
<parameter name="ReceiveAttempts" required="true">
<para>Total FAX sessions for which Asterisk is/was the recipient</para>
</parameter>
<parameter name="CompletedFAXes" required="true">
<para>Total FAX sessions which have been completed successfully</para>
</parameter>
<parameter name="FailedFAXes" required="true">
<para>Total FAX sessions which failed to complete successfully</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="SorceryMemoryCacheExpireObject" language="en_US">
<synopsis>
Expire (remove) an object from a sorcery memory cache.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Cache" required="true">
<para>The name of the cache to expire the object from.</para>
</parameter>
<parameter name="Object" required="true">
<para>The name of the object to expire.</para>
</parameter>
</syntax>
<description>
<para>Expires (removes) an object from a sorcery memory cache.</para>
</description>
</manager>
<manager name="SorceryMemoryCacheExpire" language="en_US">
<synopsis>
Expire (remove) ALL objects from a sorcery memory cache.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Cache" required="true">
<para>The name of the cache to expire all objects from.</para>
</parameter>
</syntax>
<description>
<para>Expires (removes) ALL objects from a sorcery memory cache.</para>
</description>
</manager>
<manager name="SorceryMemoryCacheStaleObject" language="en_US">
<synopsis>
Mark an object in a sorcery memory cache as stale.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Cache" required="true">
<para>The name of the cache to mark the object as stale in.</para>
</parameter>
<parameter name="Object" required="true">
<para>The name of the object to mark as stale.</para>
</parameter>
<parameter name="Reload" required="false">
<para>If true, then immediately reload the object from the backend cache instead of waiting for the next retrieval</para>
</parameter>
</syntax>
<description>
<para>Marks an object as stale within a sorcery memory cache.</para>
</description>
</manager>
<manager name="SorceryMemoryCacheStale" language="en_US">
<synopsis>
Marks ALL objects in a sorcery memory cache as stale.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Cache" required="true">
<para>The name of the cache to mark all object as stale in.</para>
</parameter>
</syntax>
<description>
<para>Marks ALL objects in a sorcery memory cache as stale.</para>
</description>
</manager>
<manager name="SorceryMemoryCachePopulate" language="en_US">
<synopsis>
Expire all objects from a memory cache and populate it with all objects from the backend.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Cache" required="true">
<para>The name of the cache to populate.</para>
</parameter>
</syntax>
<description>
<para>Expires all objects from a memory cache and populate it with all objects from the backend.</para>
</description>
</manager>
<application name="MusicOnHold" language="en_US">
<synopsis>
Play Music On Hold indefinitely.
</synopsis>
<syntax>
<parameter name="class" required="true" />
<parameter name="duration" />
</syntax>
<description>
<para>Plays hold music specified by class. If omitted, the default music
source for the channel will be used. Change the default class with
Set(CHANNEL(musicclass)=...). If duration is given, hold music will be played
specified number of seconds. If duration is ommited, music plays indefinitely.
Returns <literal>0</literal> when done, <literal>-1</literal> on hangup.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer() or Progress().</para>
</description>
</application>
<application name="StartMusicOnHold" language="en_US">
<synopsis>
Play Music On Hold.
</synopsis>
<syntax>
<parameter name="class" required="true" />
</syntax>
<description>
<para>Starts playing music on hold, uses default music class for channel.
Starts playing music specified by class. If omitted, the default music
source for the channel will be used. Always returns <literal>0</literal>.</para>
</description>
</application>
<application name="StopMusicOnHold" language="en_US">
<synopsis>
Stop playing Music On Hold.
</synopsis>
<syntax />
<description>
<para>Stops playing music on hold.</para>
</description>
</application>
<configInfo name="res_pjsip_endpoint_identifier_ip" language="en_US">
<synopsis>Module that identifies endpoints</synopsis>
<configFile name="pjsip.conf">
<configObject name="identify">
<synopsis>Identifies endpoints via some criteria.</synopsis>
<description>
<para>This module provides alternatives to matching inbound requests to
a configured endpoint. At least one of the matching mechanisms
must be provided, or the object configuration will be invalid.</para>
<para>If multiple criteria are provided, an inbound request will
be matched if it matches <emphasis>any</emphasis> of the criteria.</para>
<para>The matching mechanisms are provided by the following
configuration options:</para>
<enumlist>
<enum name="match">
<para>Match by source IP address.</para>
</enum>
<enum name="match_header">
<para>Match by SIP header.</para>
</enum>
</enumlist>
</description>
<configOption name="endpoint">
<synopsis>Name of Endpoint</synopsis>
</configOption>
<configOption name="match">
<synopsis>IP addresses or networks to match against.</synopsis>
<description>
<para>
The value is a comma-delimited list of IP addresses. IP addresses may
have a subnet mask appended. The subnet mask may be written in either
CIDR or dot-decimal notation. Separate the IP address and subnet
mask with a slash ('/').
</para>
</description>
</configOption>
<configOption name="srv_lookups" default="yes">
<synopsis>Perform SRV lookups for provided hostnames.</synopsis>
<description>
<para>When enabled, <replaceable>srv_lookups</replaceable> will
perform SRV lookups for _sip._udp, _sip._tcp, and _sips._tcp of the given
hostnames to determine additional addresses that traffic may originate from.
</para>
</description>
</configOption>
<configOption name="match_header">
<synopsis>Header/value pair to match against.</synopsis>
<description>
<para>A SIP header who value is used to match against. SIP
requests containing the header, along with the specified value, will be
mapped to the specified endpoint. The header must be specified with a
<literal>:</literal>, as in <literal>match_header = SIPHeader: value</literal>.
</para>
</description>
</configOption>
<configOption name="type">
<synopsis>Must be of type 'identify'.</synopsis>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="PJSIPShowRegistrationsInbound" language="en_US">
<synopsis>
Lists PJSIP inbound registrations.
</synopsis>
<syntax />
<description>
<para>
In response, <literal>InboundRegistrationDetail</literal> events showing configuration
and status information are raised for all contacts, static or dynamic. Once all events
are completed an <literal>InboundRegistrationDetailComplete</literal> is issued.
</para>
<warning>
<para>
This command just dumps all coonfigured AORs with contacts, even if the contact
is a permanent one. To really get just inbound registrations, use
<literal>PJSIPShowRegistrationInboundContactStatuses</literal>.
</para>
</warning>
</description>
<see-also>
<ref type="manager" module="res_pjsip_registrar">PJSIPShowRegistrationInboundContactStatuses</ref>
</see-also>
</manager>
<manager name="PJSIPShowRegistrationInboundContactStatuses" language="en_US">
<synopsis>
Lists ContactStatuses for PJSIP inbound registrations.
</synopsis>
<syntax />
<description>
<para>
In response, <literal>ContactStatusDetail</literal> events showing status information
are raised for each inbound registration (dynamic contact) object. Once all events
are completed a <literal>ContactStatusDetailComplete</literal> event is issued.
</para>
</description>
</manager>
<configInfo name="res_pjsip_config_wizard" language="en_US">
<synopsis>Module that privides simple configuration wizard capabilities.</synopsis>
<description>
<para>
<emphasis>PJSIP Configuration Wizard</emphasis>
</para>
<para>This module allows creation of common PJSIP configuration scenarios
without having to specify individual endpoint, aor, auth, identify and registration objects.
</para>
<para> </para>
<para>For example, the following configuration snippet would create the
endpoint, aor, contact, auth and phoneprov objects necessary for a phone to
get phone provisioning information, register, and make and receive calls.
A hint is also created in the default context for extension 1000.</para>
<para> </para>
<para>[myphone]</para>
<para>type = wizard</para>
<para>sends_auth = no</para>
<para>accepts_auth = yes</para>
<para>sends_registrations = no</para>
<para>accepts_registrations = yes</para>
<para>has_phoneprov = yes</para>
<para>transport = ipv4</para>
<para>has_hint = yes</para>
<para>hint_exten = 1000</para>
<para>inbound_auth/username = testname</para>
<para>inbound_auth/password = test password</para>
<para>endpoint/allow = ulaw</para>
<para>endpoint/context = default</para>
<para>phoneprov/MAC = 001122aa4455</para>
<para>phoneprov/PROFILE = profile1</para>
<para> </para>
<para>The first 8 items are specific to the wizard. The rest of the items
are passed verbatim to the underlying objects.</para>
<para> </para>
<para>The following configuration snippet would create the
endpoint, aor, contact, auth, identify and registration objects necessary for a trunk
to another pbx or ITSP that requires registration.</para>
<para> </para>
<para>[mytrunk]</para>
<para>type = wizard</para>
<para>sends_auth = yes</para>
<para>accepts_auth = no</para>
<para>sends_registrations = yes</para>
<para>accepts_registrations = no</para>
<para>transport = ipv4</para>
<para>remote_hosts = sip1.myitsp.com:5060,sip2.myitsp.com:5060</para>
<para>outbound_auth/username = testname</para>
<para>outbound_auth/password = test password</para>
<para>endpoint/allow = ulaw</para>
<para>endpoint/context = default</para>
<para> </para>
<para>Of course, any of the items in either example could be placed into
templates and shared among wizard objects.</para>
<para> </para>
<para>For more information, visit:</para>
<para>
<literal>https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard</literal>
</para>
</description>
<configFile name="pjsip_wizard.conf">
<configObject name="wizard">
<synopsis>Provides config wizard.</synopsis>
<description>
<para>For more information, visit:</para>
<para>
<literal>https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard</literal>
</para>
</description>
<configOption name="type">
<synopsis>Must be 'wizard'.</synopsis>
</configOption>
<configOption name="transport">
<synopsis>The name of a transport to use for this object.</synopsis>
<description>
<para>If not specified,
the default will be used.</para>
</description>
</configOption>
<configOption name="remote_hosts">
<synopsis>List of remote hosts.</synopsis>
<description>
<para>A comma-separated list of remote hosts in the form of
<replaceable>host</replaceable>[:<replaceable>port</replaceable>].
If set, an aor static contact and an identify match will be created for each
entry in the list. If send_registrations is also set, a registration will
also be created for each.</para>
</description>
</configOption>
<configOption name="outbound_proxy">
<synopsis>Shortcut for specifying proxy on individual objects.</synopsis>
<description>
<para>Shortcut for specifying endpoint/outbound_proxy,
aor/outbound_proxy, and registration/outbound_proxy individually.
</para>
</description>
</configOption>
<configOption name="sends_auth" default="no">
<synopsis>Send outbound authentication to remote hosts.</synopsis>
<description>
<para>At least outbound_auth/username is required.</para>
</description>
</configOption>
<configOption name="accepts_auth" default="no">
<synopsis>Accept incoming authentication from remote hosts.</synopsis>
<description>
<para>At least inbound_auth/username is required.</para>
</description>
</configOption>
<configOption name="sends_registrations" default="no">
<synopsis>Send outbound registrations to remote hosts.</synopsis>
<description>
<para>remote_hosts is required and a registration object will
be created for each host in the remote _hosts string. If authentication is required,
sends_auth and an outbound_auth/username must also be supplied.</para>
</description>
</configOption>
<configOption name="sends_line_with_registrations" default="no">
<synopsis>Sets "line" and "endpoint parameters on registrations.</synopsis>
<description>
<para>Setting this to true will cause the wizard to skip the
creation of an identify object to match incoming requests to the endpoint and
instead add the line and endpoint parameters to the outbound registration object.
</para>
</description>
</configOption>
<configOption name="accepts_registrations" default="no">
<synopsis>Accept inbound registration from remote hosts.</synopsis>
<description>
<para>An AOR with dynamic contacts will be created. If
the number of contacts nneds to be limited, set aor/max_contacts.</para>
</description>
</configOption>
<configOption name="has_phoneprov" default="no">
<synopsis>Create a phoneprov object for this endpoint.</synopsis>
<description>
<para>A phoneprov object will be created. phoneprov/MAC
must be specified.</para>
</description>
</configOption>
<configOption name="server_uri_pattern" default="sip:${REMOTE_HOST}">
<synopsis>A pattern to use for constructing outbound registration server_uris.</synopsis>
<description>
<para>
The literal <literal>${REMOTE_HOST}</literal> will be substituted with the
appropriate remote_host for each registration.</para>
</description>
</configOption>
<configOption name="client_uri_pattern" default="sip:${USERNAME}@${REMOTE_HOST}">
<synopsis>A pattern to use for constructing outbound registration client_uris.</synopsis>
<description>
<para>
The literals <literal>${REMOTE_HOST}</literal> and <literal>${USERNAME}</literal>
will be substituted with the appropriate remote_host and outbound_auth/username.</para>
</description>
</configOption>
<configOption name="contact_pattern" default="sip:${REMOTE_HOST}">
<synopsis>A pattern to use for constructing outbound contact uris.</synopsis>
<description>
<para>
The literal <literal>${REMOTE_HOST}</literal> will be substituted with the
appropriate remote_host for each contact.</para>
</description>
</configOption>
<configOption name="has_hint" default="no">
<synopsis>Create hint and optionally a default application.</synopsis>
<description>
<para>Create hint and optionally a default application.</para>
</description>
</configOption>
<configOption name="hint_context" default="endpoint/context or 'default'">
<synopsis>The context in which to place hints.</synopsis>
<description>
<para>Ignored if <literal>hint_exten</literal> is not specified otherwise specifies the
context into which the dialplan hints will be placed. If not specified,
defaults to the endpoint's context or <literal>default</literal> if that isn't
found.
</para>
</description>
</configOption>
<configOption name="hint_exten">
<synopsis>Extension to map a PJSIP hint to.</synopsis>
<description>
<para>Will create the following entry in <literal>hint_context</literal>:</para>
<para>
<literal>exten =&gt; &lt;hint_exten&gt;,hint,PJSIP/&lt;wizard_id&gt;</literal>
</para>
<para> </para>
<para>Normal dialplan precedence rules apply so if there's already a hint for
this extension in <literal>hint_context</literal>, this one will be ignored.
For more information, visit: </para>
<para>
<literal>https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard</literal>
</para>
</description>
</configOption>
<configOption name="hint_application">
<synopsis>Application to call when 'hint_exten' is dialed.</synopsis>
<description>
<para>Ignored if <literal>hint_exten</literal> isn't specified otherwise
will create the following priority 1 extension in <literal>hint_context</literal>:</para>
<para>
<literal>exten =&gt; &lt;hint_exten&gt;,1,&lt;hint_application&gt;</literal>
</para>
<para> </para>
<para>You can specify any valid extensions.conf application expression.</para>
<para>Examples: </para>
<para>
<literal>Dial(${HINT})</literal>
</para>
<para>
<literal>Gosub(stdexten,${EXTEN},1(${HINT}))</literal>
</para>
<para> </para>
<para>Any extensions.conf style variables specified are passed directly to the
dialplan.</para>
<para> </para>
<para>Normal dialplan precedence rules apply so if there's already a priority 1
application for this specific extension in <literal>hint_context</literal>,
this one will be ignored. For more information, visit: </para>
<para>
<literal>https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard</literal>
</para>
</description>
</configOption>
<configOption name="endpoint&#47;*">
<synopsis>Variables to be passed directly to the endpoint.</synopsis>
</configOption>
<configOption name="aor&#47;*">
<synopsis>Variables to be passed directly to the aor.</synopsis>
<description>
<para>If an aor/contact is explicitly defined then remote_hosts
will not be used to create contacts automatically.</para>
</description>
</configOption>
<configOption name="inbound_auth&#47;*">
<synopsis>Variables to be passed directly to the inbound auth.</synopsis>
</configOption>
<configOption name="outbound_auth&#47;*">
<synopsis>Variables to be passed directly to the outbound auth.</synopsis>
</configOption>
<configOption name="identify&#47;*">
<synopsis>Variables to be passed directly to the identify.</synopsis>
<description>
<para>If an identify/match is explicitly defined then remote_hosts
will not be used to create matches automatically.</para>
</description>
</configOption>
<configOption name="registration&#47;*">
<synopsis>Variables to be passed directly to the outbound registrations.</synopsis>
</configOption>
<configOption name="phoneprov&#47;*">
<synopsis>Variables to be passed directly to the phoneprov object.</synopsis>
<description>
<para>
To activate phoneprov, at least phoneprov/MAC must be set.</para>
</description>
</configOption>
</configObject>
</configFile>
</configInfo>
<manager name="MWIGet" language="en_US">
<synopsis>
Get selected mailboxes with message counts.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Mailbox ID in the form of
/<replaceable>regex</replaceable>/ for all mailboxes matching the regular
expression. Otherwise it is for a specific mailbox.</para>
</parameter>
</syntax>
<description>
<para>Get a list of mailboxes with their message counts.</para>
</description>
</manager>
<managerEvent language="en_US" name="MWIGet">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>
Raised in response to a MWIGet command.
</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="Mailbox">
<para>Specific mailbox ID.</para>
</parameter>
<parameter name="OldMessages">
<para>The number of old messages in the mailbox.</para>
</parameter>
<parameter name="NewMessages">
<para>The number of new messages in the mailbox.</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">MWIGet</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MWIGetComplete">
<managerEventInstance class="EVENT_FLAG_REPORTING">
<synopsis>
Raised in response to a MWIGet command.
</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="EventList" />
<parameter name="ListItems">
<para>The number of mailboxes reported.</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">MWIGet</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="MWIDelete" language="en_US">
<synopsis>
Delete selected mailboxes.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<xi:include xpointer="xpointer(/docs/manager[@name='MWIGet']/syntax/parameter[@name='Mailbox'])" />
</syntax>
<description>
<para>Delete the specified mailboxes.</para>
</description>
</manager>
<manager name="MWIUpdate" language="en_US">
<synopsis>
Update the mailbox message counts.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Specific mailbox ID.</para>
</parameter>
<parameter name="OldMessages">
<para>The number of old messages in the mailbox. Defaults
to zero if missing.</para>
</parameter>
<parameter name="NewMessages">
<para>The number of new messages in the mailbox. Defaults
to zero if missing.</para>
</parameter>
</syntax>
<description>
<para>Update the mailbox message counts.</para>
</description>
</manager>
<manager name="DeviceStateList" language="en_US">
<synopsis>
List the current known device states.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>This will list out all known device states in a
sequence of <replaceable>DeviceStateChange</replaceable> events.
When finished, a <replaceable>DeviceStateListComplete</replaceable> event
will be emitted.</para>
</description>
<see-also>
<ref type="managerEvent">DeviceStateChange</ref>
<ref type="function">DEVICE_STATE</ref>
</see-also>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='DeviceStateChange'])" />
</list-elements>
<managerEvent name="DeviceStateListComplete" language="en_US">
<managerEventInstance class="EVENT_FLAG_COMMAND">
<synopsis>
Indicates the end of the list the current known extension states.
</synopsis>
<syntax>
<parameter name="EventList">
<para>Conveys the status of the event list.</para>
</parameter>
<parameter name="ListItems">
<para>Conveys the number of statuses reported.</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
</responses>
</manager>
<info name="CHANNEL" language="en_US" tech="OOH323">
<enumlist>
<enum name="faxdetect">
<para>R/W Fax Detect</para>
<para>Returns 0 or 1</para>
<para>Write yes or no</para>
</enum>
<enum name="t38support">
<para>R/W t38support</para>
<para>Returns 0 or 1</para>
<para>Write yes or no</para>
</enum>
<enum name="h323id_url">
<para>R/0 Returns caller URL</para>
</enum>
<enum name="caller_h323id">
<para>R/0 Returns caller h323id</para>
</enum>
<enum name="caller_dialeddigits">
<para>R/0 Returns caller dialed digits</para>
</enum>
<enum name="caller_email">
<para>R/0 Returns caller email</para>
</enum>
<enum name="callee_email">
<para>R/0 Returns callee email</para>
</enum>
<enum name="callee_dialeddigits">
<para>R/0 Returns callee dialed digits</para>
</enum>
<enum name="caller_url">
<para>R/0 Returns caller URL</para>
</enum>
<enum name="max_forwards">
<para>R/W Get or set the maximum number of call forwards for this channel.
This number describes the number of times a call may be forwarded by this channel
before the call fails. "Forwards" in this case refers to redirects by phones as well
as calls to local channels.
Note that this has no relation to the SIP Max-Forwards header.
</para>
</enum>
</enumlist>
</info>
</docs>