2467 lines
114 KiB
XML
2467 lines
114 KiB
XML
<?xml version="1.0" encoding="UTF-8"?>
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<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
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<?xml-stylesheet type="text/xsl" href="appdocsxml.xslt"?>
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<docs xmlns:xi="http://www.w3.org/2001/XInclude">
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<configInfo name="res_pjsip" language="en_US">
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<synopsis>SIP Resource using PJProject</synopsis>
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<configFile name="pjsip.conf">
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<configObject name="endpoint">
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<synopsis>Endpoint</synopsis>
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<description><para>
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The <emphasis>Endpoint</emphasis> is the primary configuration object.
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It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
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dialable entries of their own. Communication with another SIP device is
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accomplished via Addresses of Record (AoRs) which have one or more
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contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to
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use a <literal>transport</literal> will default to first transport found
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in <filename>pjsip.conf</filename> that matches its type.
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</para>
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<para>Example: An Endpoint has been configured with no transport.
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When it comes time to call an AoR, PJSIP will find the
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first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
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will use the first IPv6 transport and try to send the request.
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</para>
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<para>If the anonymous endpoint identifier is in use an endpoint with the name
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"anonymous@domain" will be searched for as a last resort. If this is not found
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it will fall back to searching for "anonymous". If neither endpoints are found
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the anonymous endpoint identifier will not return an endpoint and anonymous
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calling will not be possible.
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</para>
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</description>
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<configOption name="100rel" default="yes">
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<synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
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<description>
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<enumlist>
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<enum name="no">
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<para>If set to <literal>no</literal>, do not support transmission of
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reliable provisional responses. As UAS, if an incoming request contains 100rel
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in the Required header, it is rejected with 420 Bad Extension.</para>
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</enum>
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<enum name="required">
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<para>If set to <literal>required</literal>, require provisional responses to
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be sent and received reliably. As UAS, incoming requests without 100rel
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in the Supported header are rejected with 421 Extension Required. As UAC,
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outgoing requests will have 100rel in the Required header.</para>
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</enum>
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<enum name="peer_supported">
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<para>If set to <literal>peer_supported</literal>, send provisional responses
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reliably if the request by the peer contained 100rel in the Supported or
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Require header. As UAS, if an incoming request contains 100rel in the Supported
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header, send 1xx responses reliably. If the request by the peer does not contain 100rel
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in the Supported and Require header, send responses normally. As UAC, outgoing requests
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will contain 100rel in the Supported header.</para>
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</enum>
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<enum name="yes">
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<para>If set to <literal>yes</literal>, indicate the support of reliable provisional
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responses and PRACK them if required by the peer. As UAS, if the incoming request
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contains 100rel in the Supported header but not in the Required header, send 1xx
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responses normally. If the incoming request contains 100rel in the Required header,
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send 1xx responses reliably. As UAC add 100rel to the Supported header and PRACK 1xx
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responses if required.</para>
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</enum>
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</enumlist>
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</description>
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</configOption>
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<configOption name="aggregate_mwi" default="yes">
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<synopsis>Condense MWI notifications into a single NOTIFY.</synopsis>
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<description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
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waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
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individual NOTIFYs are sent for each mailbox.</para></description>
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</configOption>
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<configOption name="allow">
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<synopsis>Media Codec(s) to allow</synopsis>
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</configOption>
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<configOption name="codec_prefs_incoming_offer">
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<synopsis>Codec negotiation prefs for incoming offers.</synopsis>
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<description>
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<para>
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This is a string that describes how the codecs
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specified on an incoming SDP offer (pending) are reconciled with the codecs specified
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on an endpoint (configured) before being sent to the Asterisk core.
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The string actually specifies 4 <literal>name:value</literal> pair parameters
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separated by commas. Whitespace is ignored and they may be specified in any order.
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Note that this option is reserved for future functionality.
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</para>
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<para>
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Parameters:
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</para>
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<enumlist>
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<enum name="prefer: < pending | configured >">
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<para>
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</para>
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<enumlist>
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<enum name="pending"><para>The codec list from the caller. (default)</para></enum>
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<enum name="configured"><para>The codec list from the endpoint.</para></enum>
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</enumlist>
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</enum>
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<enum name="operation : < intersect | only_preferred | only_nonpreferred >">
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<para>
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</para>
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<enumlist>
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<enum name="intersect"><para>Only common codecs with the preferred codecs first. (default)</para></enum>
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<enum name="only_preferred"><para>Use only the preferred codecs.</para></enum>
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<enum name="only_nonpreferred"><para>Use only the non-preferred codecs.</para></enum>
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</enumlist>
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</enum>
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<enum name="keep : < all | first >">
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<para>
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</para>
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<enumlist>
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<enum name="all"><para>After the operation, keep all codecs. (default)</para></enum>
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<enum name="first"><para>After the operation, keep only the first codec.</para></enum>
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</enumlist>
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</enum>
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<enum name="transcode : < allow | prevent >">
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<para>
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</para>
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<enumlist>
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<enum name="allow"><para>Allow transcoding. (default)</para></enum>
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<enum name="prevent"><para>Prevent transcoding.</para></enum>
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</enumlist>
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</enum>
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</enumlist>
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<para>
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</para>
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<example>
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codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: all, transcode: allow
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</example>
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<para>
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Prefer the codecs coming from the caller. Use only the ones that are common.
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keeping the order of the preferred list. Keep all codecs in the result. Allow transcoding.
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</para>
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</description>
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</configOption>
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<configOption name="codec_prefs_outgoing_offer">
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<synopsis>Codec negotiation prefs for outgoing offers.</synopsis>
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<description>
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<para>
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This is a string that describes how the codecs specified in the topology that
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comes from the Asterisk core (pending) are reconciled with the codecs specified on an
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endpoint (configured) when sending an SDP offer.
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The string actually specifies 4 <literal>name:value</literal> pair parameters
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separated by commas. Whitespace is ignored and they may be specified in any order.
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Note that this option is reserved for future functionality.
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</para>
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<para>
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Parameters:
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</para>
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<enumlist>
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<enum name="prefer: < pending | configured >">
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<para>
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</para>
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<enumlist>
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<enum name="pending"><para>The codec list from the core. (default)</para></enum>
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<enum name="configured"><para>The codec list from the endpoint.</para></enum>
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</enumlist>
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</enum>
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<enum name="operation : < union | intersect | only_preferred | only_nonpreferred >">
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<para>
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</para>
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<enumlist>
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<enum name="union"><para>Merge the lists with the preferred codecs first. (default)</para></enum>
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<enum name="intersect"><para>Only common codecs with the preferred codecs first. (default)</para></enum>
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<enum name="only_preferred"><para>Use only the preferred codecs.</para></enum>
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<enum name="only_nonpreferred"><para>Use only the non-preferred codecs.</para></enum>
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</enumlist>
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</enum>
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<enum name="keep : < all | first >">
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<para>
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</para>
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<enumlist>
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<enum name="all"><para>After the operation, keep all codecs. (default)</para></enum>
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<enum name="first"><para>After the operation, keep only the first codec.</para></enum>
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</enumlist>
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</enum>
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<enum name="transcode : < allow | prevent >">
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<para>
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</para>
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<enumlist>
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<enum name="allow"><para>Allow transcoding. (default)</para></enum>
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<enum name="prevent"><para>Prevent transcoding.</para></enum>
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</enumlist>
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</enum>
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</enumlist>
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<para>
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</para>
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<example>
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codec_prefs_outgoing_offer = prefer: configured, operation: union, keep: first, transcode: prevent
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</example>
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<para>
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Prefer the codecs coming from the endpoint. Merge them with the codecs from the core
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keeping the order of the preferred list. Keep only the first one. No transcoding allowed.
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</para>
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</description>
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</configOption>
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<configOption name="codec_prefs_incoming_answer">
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<synopsis>Codec negotiation prefs for incoming answers.</synopsis>
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<description>
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<para>
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This is a string that describes how the codecs specified in an incoming SDP answer
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(pending) are reconciled with the codecs specified on an endpoint (configured)
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when receiving an SDP answer.
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The string actually specifies 4 <literal>name:value</literal> pair parameters
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separated by commas. Whitespace is ignored and they may be specified in any order.
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Note that this option is reserved for future functionality.
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</para>
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<para>
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Parameters:
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</para>
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<enumlist>
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<enum name="prefer: < pending | configured >">
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<para>
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</para>
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<enumlist>
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<enum name="pending"><para>The codec list in the received SDP answer. (default)</para></enum>
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<enum name="configured"><para>The codec list from the endpoint.</para></enum>
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</enumlist>
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</enum>
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<enum name="operation : < union | intersect | only_preferred | only_nonpreferred >">
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<para>
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</para>
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<enumlist>
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<enum name="union"><para>Merge the lists with the preferred codecs first.</para></enum>
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<enum name="intersect"><para>Only common codecs with the preferred codecs first. (default)</para></enum>
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<enum name="only_preferred"><para>Use only the preferred codecs.</para></enum>
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<enum name="only_nonpreferred"><para>Use only the non-preferred codecs.</para></enum>
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</enumlist>
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</enum>
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<enum name="keep : < all | first >">
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<para>
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</para>
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<enumlist>
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<enum name="all"><para>After the operation, keep all codecs. (default)</para></enum>
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<enum name="first"><para>After the operation, keep only the first codec.</para></enum>
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</enumlist>
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</enum>
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<enum name="transcode : < allow | prevent >">
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<para>
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The transcode parameter is ignored when processing answers.
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</para>
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</enum>
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</enumlist>
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<para>
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</para>
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<example>
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codec_prefs_incoming_answer = keep: first
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</example>
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<para>
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Use the defaults but keep oinly the first codec.
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</para>
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</description>
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</configOption>
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<configOption name="codec_prefs_outgoing_answer">
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<synopsis>Codec negotiation prefs for outgoing answers.</synopsis>
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<description>
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<para>
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This is a string that describes how the codecs that come from the core (pending)
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are reconciled with the codecs specified on an endpoint (configured)
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when sending an SDP answer.
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The string actually specifies 4 <literal>name:value</literal> pair parameters
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separated by commas. Whitespace is ignored and they may be specified in any order.
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Note that this option is reserved for future functionality.
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</para>
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<para>
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Parameters:
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</para>
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<enumlist>
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<enum name="prefer: < pending | configured >">
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<para>
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</para>
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<enumlist>
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<enum name="pending"><para>The codec list that came from the core. (default)</para></enum>
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<enum name="configured"><para>The codec list from the endpoint.</para></enum>
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</enumlist>
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</enum>
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<enum name="operation : < union | intersect | only_preferred | only_nonpreferred >">
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<para>
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</para>
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<enumlist>
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<enum name="union"><para>Merge the lists with the preferred codecs first.</para></enum>
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<enum name="intersect"><para>Only common codecs with the preferred codecs first. (default)</para></enum>
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<enum name="only_preferred"><para>Use only the preferred codecs.</para></enum>
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<enum name="only_nonpreferred"><para>Use only the non-preferred codecs.</para></enum>
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</enumlist>
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</enum>
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<enum name="keep : < all | first >">
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<para>
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</para>
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<enumlist>
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<enum name="all"><para>After the operation, keep all codecs. (default)</para></enum>
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<enum name="first"><para>After the operation, keep only the first codec.</para></enum>
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</enumlist>
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</enum>
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<enum name="transcode : < allow | prevent >">
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<para>
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The transcode parameter is ignored when processing answers.
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</para>
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</enum>
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</enumlist>
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<para>
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</para>
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<example>
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codec_prefs_incoming_answer = keep: first
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</example>
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<para>
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Use the defaults but keep oinly the first codec.
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</para>
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</description>
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</configOption>
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<configOption name="allow_overlap" default="yes">
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<synopsis>Enable RFC3578 overlap dialing support.</synopsis>
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</configOption>
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<configOption name="overlap_context">
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<synopsis>Dialplan context to use for RFC3578 overlap dialing.</synopsis>
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<description>
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<para>Dialplan context to use for overlap dialing extension matching.
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If not specified, the context configured for the endpoint will be used.
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If specified, the extensions/patterns in the specified context will be used
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for determining if a full number has been received from the endpoint.
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</para>
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</description>
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</configOption>
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<configOption name="aors">
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<synopsis>AoR(s) to be used with the endpoint</synopsis>
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<description><para>
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List of comma separated AoRs that the endpoint should be associated with.
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</para></description>
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</configOption>
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<configOption name="auth">
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<synopsis>Authentication Object(s) associated with the endpoint</synopsis>
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<description><para>
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This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
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in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
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</para><para>
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Endpoints without an authentication object
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configured will allow connections without verification.</para>
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<note><para>
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Using the same auth section for inbound and outbound
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authentication is not recommended. There is a difference in
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meaning for an empty realm setting between inbound and outbound
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authentication uses. See the auth realm description for details.
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</para></note>
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</description>
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</configOption>
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<configOption name="callerid">
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<synopsis>CallerID information for the endpoint</synopsis>
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<description><para>
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Must be in the format <literal>Name <Number></literal>,
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or only <literal><Number></literal>.
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</para></description>
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</configOption>
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<configOption name="callerid_privacy">
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<synopsis>Default privacy level</synopsis>
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<description>
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<enumlist>
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<enum name="allowed_not_screened" />
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<enum name="allowed_passed_screen" />
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<enum name="allowed_failed_screen" />
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<enum name="allowed" />
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<enum name="prohib_not_screened" />
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<enum name="prohib_passed_screen" />
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<enum name="prohib_failed_screen" />
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<enum name="prohib" />
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<enum name="unavailable" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="callerid_tag">
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<synopsis>Internal id_tag for the endpoint</synopsis>
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</configOption>
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<configOption name="context">
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<synopsis>Dialplan context for inbound sessions</synopsis>
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</configOption>
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<configOption name="direct_media_glare_mitigation" default="none">
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<synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
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|
<description>
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<para>
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This setting attempts to avoid creating INVITE glare scenarios
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by disabling direct media reINVITEs in one direction thereby allowing
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designated servers (according to this option) to initiate direct
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media reINVITEs without contention and significantly reducing call
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setup time.
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</para>
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<para>
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A more detailed description of how this option functions can be found in
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the Asterisk documentation https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Concepts/SIP-Direct-Media-Reinvite-Glare-Avoidance/
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</para>
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<enumlist>
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<enum name="none" />
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|
<enum name="outgoing" />
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|
<enum name="incoming" />
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|
</enumlist>
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</description>
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</configOption>
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<configOption name="direct_media_method" default="invite">
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<synopsis>Direct Media method type</synopsis>
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|
<description>
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<para>Method for setting up Direct Media between endpoints.</para>
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|
<enumlist>
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<enum name="invite" />
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|
<enum name="reinvite">
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<para>Alias for the <literal>invite</literal> value.</para>
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</enum>
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|
<enum name="update" />
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|
</enumlist>
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|
</description>
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</configOption>
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<configOption name="trust_connected_line">
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<synopsis>Accept Connected Line updates from this endpoint</synopsis>
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|
</configOption>
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<configOption name="send_connected_line">
|
|
<synopsis>Send Connected Line updates to this endpoint</synopsis>
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|
</configOption>
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|
<configOption name="connected_line_method" default="invite">
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|
<synopsis>Connected line method type</synopsis>
|
|
<description>
|
|
<para>Method used when updating connected line information.</para>
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|
<enumlist>
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|
<enum name="invite">
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|
<para>When set to <literal>invite</literal>, check the remote's Allow header and
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if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP
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renegotiation. If UPDATE is not Allowed, send INVITE.</para>
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</enum>
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|
<enum name="reinvite">
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|
<para>Alias for the <literal>invite</literal> value.</para>
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</enum>
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<enum name="update">
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<para>If set to <literal>update</literal>, send UPDATE regardless of what the remote
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Allows. </para>
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</enum>
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|
</enumlist>
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|
</description>
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</configOption>
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<configOption name="direct_media" default="yes">
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|
<synopsis>Determines whether media may flow directly between endpoints.</synopsis>
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|
</configOption>
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|
<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">
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|
<para>DTMF is sent out of band of the main audio stream.</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>
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|
</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">
|
|
<synopsis>Way(s) for the endpoint to be identified</synopsis>
|
|
<description>
|
|
<para>Endpoints and AORs can be identified in multiple ways. This
|
|
option is a comma separated list of methods the endpoint can be
|
|
identified.
|
|
</para>
|
|
<note><para>
|
|
This option controls both how an endpoint is matched for incoming
|
|
traffic and also how an AOR is determined if a registration
|
|
occurs. You must list at least one method that also matches for
|
|
AORs or the registration will fail.
|
|
</para></note>
|
|
<enumlist>
|
|
<enum name="username">
|
|
<para>Matches the endpoint or AOR ID based on the username
|
|
and domain in the From header (or To header for AORs). If
|
|
an exact match on both username and domain/realm fails, the
|
|
match is retried with just the username.
|
|
</para>
|
|
</enum>
|
|
<enum name="auth_username">
|
|
<para>Matches the endpoint or AOR ID based on the username
|
|
and realm in the Authentication header. If an exact match
|
|
on both username and domain/realm fails, the match is
|
|
retried with just the username.
|
|
</para>
|
|
<note><para>This method of identification 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 this method
|
|
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 <literal>unidentified_request</literal>
|
|
parameters in the "global" configuration object.
|
|
</para></note>
|
|
</enum>
|
|
<enum name="ip">
|
|
<para>Matches the endpoint based on the source IP address.
|
|
</para>
|
|
<para>This method of identification is not configured here
|
|
but simply allowed by this configuration option. See the
|
|
documentation for the <literal>identify</literal>
|
|
configuration section for more details on this method of
|
|
endpoint identification.
|
|
</para>
|
|
</enum>
|
|
<enum name="header">
|
|
<para>Matches the endpoint based on a configured SIP header
|
|
value.
|
|
</para>
|
|
<para>This method of identification is not configured here
|
|
but simply allowed by this configuration option. See the
|
|
documentation for the <literal>identify</literal>
|
|
configuration section for more details on this method of
|
|
endpoint identification.
|
|
</para>
|
|
</enum>
|
|
</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_mwi_external 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_history_info" default="no">
|
|
<synopsis>Send the History-Info header, conveying the diversion
|
|
information to the called and calling user agents</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>
|
|
Minimum 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>
|
|
Maximum session timer expiration period. Time in seconds.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="transport">
|
|
<synopsis>Explicit transport configuration to use</synopsis>
|
|
<description>
|
|
<para>This will <emphasis>force</emphasis> the endpoint to use the
|
|
specified transport configuration to send SIP messages. You need
|
|
to already know what kind of transport (UDP/TCP/IPv4/etc) the
|
|
endpoint device will use.
|
|
</para>
|
|
<note><para>Not specifying a transport will select the first
|
|
configured transport in <filename>pjsip.conf</filename> which is
|
|
compatible with the URI we are trying to contact.
|
|
</para></note>
|
|
<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 packetization 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>
|
|
Redundancy 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="t38_bind_udptl_to_media_address" default="no">
|
|
<synopsis>Bind the UDPTL instance to the media_adress</synopsis>
|
|
<description><para>
|
|
If media_address is specified, this option causes the UDPTL instance to be bound to
|
|
the specified ip address which causes the packets to be sent from that address.
|
|
</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="moh_passthrough" default="no">
|
|
<synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</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://docs.asterisk.org/Configuration/Channel-Drivers/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://docs.asterisk.org/Configuration/Channel-Drivers/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://docs.asterisk.org/Configuration/Channel-Drivers/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://docs.asterisk.org/Configuration/Channel-Drivers/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 use 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><para>
|
|
It can be one of the following values:
|
|
</para><enumlist>
|
|
<enum name="no"><para>
|
|
meaning no verification is done.
|
|
</para></enum>
|
|
<enum name="fingerprint"><para>
|
|
meaning to verify the remote fingerprint.
|
|
</para></enum>
|
|
<enum name="certificate"><para>
|
|
meaning to verify the remote certificate.
|
|
</para></enum>
|
|
<enum name="yes"><para>
|
|
meaning to verify both the remote fingerprint and certificate.
|
|
</para></enum>
|
|
</enumlist>
|
|
</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_auto_generate_cert" default="no">
|
|
<synopsis>Whether or not to automatically generate an ephemeral X.509 certificate</synopsis>
|
|
<description>
|
|
<para>
|
|
If enabled, Asterisk will generate an X.509 certificate for each DTLS session.
|
|
This option only applies if <replaceable>media_encryption</replaceable> is set
|
|
to <literal>dtls</literal>. This option will be automatically enabled if
|
|
<literal>webrtc</literal> is enabled and <literal>dtls_cert_file</literal> is
|
|
not specified.
|
|
</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="preferred_codec_only" default="no">
|
|
<synopsis>Respond to a SIP invite with the single most preferred codec (DEPRECATED)</synopsis>
|
|
<description><para>Respond to a SIP invite with the single most preferred codec
|
|
rather than advertising all joint codec capabilities. This limits the other side's codec
|
|
choice to exactly what we prefer.</para>
|
|
<warning><para>This option has been deprecated in favor of
|
|
<literal>incoming_call_offer_pref</literal>. Setting both options is unsupported.</para>
|
|
</warning>
|
|
</description>
|
|
<see-also>
|
|
<ref type="configOption">incoming_call_offer_pref</ref>
|
|
</see-also>
|
|
</configOption>
|
|
<configOption name="incoming_call_offer_pref" default="local">
|
|
<synopsis>Preferences for selecting codecs for an incoming call.</synopsis>
|
|
<description>
|
|
<para>Based on this setting, a joint list of preferred codecs between those
|
|
received in an incoming SDP offer (remote), and those specified in the
|
|
endpoint's "allow" parameter (local) es created and is passed to the Asterisk
|
|
core. </para>
|
|
<note><para>This list will consist of only those codecs found in both lists.</para></note>
|
|
<enumlist>
|
|
<enum name="local"><para>
|
|
Include all codecs in the local list that are also in the remote list
|
|
preserving the local order. (default).
|
|
</para></enum>
|
|
<enum name="local_first"><para>
|
|
Include only the first codec in the local list that is also in the remote list.
|
|
</para></enum>
|
|
<enum name="remote"><para>
|
|
Include all codecs in the remote list that are also in the local list
|
|
preserving the remote order.
|
|
</para></enum>
|
|
<enum name="remote_first"><para>
|
|
Include only the first codec in the remote list that is also in the local list.
|
|
</para></enum>
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="outgoing_call_offer_pref" default="remote_merge">
|
|
<synopsis>Preferences for selecting codecs for an outgoing call.</synopsis>
|
|
<description>
|
|
<para>Based on this setting, a joint list of preferred codecs between
|
|
those received from the Asterisk core (remote), and those specified in
|
|
the endpoint's "allow" parameter (local) is created and is used to create
|
|
the outgoing SDP offer.</para>
|
|
<enumlist>
|
|
<enum name="local"><para>
|
|
Include all codecs in the local list that are also in the remote list
|
|
preserving the local order.
|
|
</para></enum>
|
|
<enum name="local_merge"><para>
|
|
Include all codecs in the local list preserving the local order.
|
|
</para></enum>
|
|
<enum name="local_first"><para>
|
|
Include only the first codec in the local list.
|
|
</para></enum>
|
|
<enum name="remote"><para>
|
|
Include all codecs in the remote list that are also in the local list
|
|
preserving the remote order.
|
|
</para></enum>
|
|
<enum name="remote_merge"><para>
|
|
Include all codecs in the local list preserving the remote order. (default)
|
|
</para></enum>
|
|
<enum name="remote_first"><para>
|
|
Include only the first codec in the remote list that is also in the local list.
|
|
</para></enum>
|
|
</enumlist>
|
|
</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&Ringing state</synopsis>
|
|
<description><para>
|
|
Control whether dialog-info subscriptions get 'early' state
|
|
on Ringing when already INUSE.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="max_audio_streams" default="1">
|
|
<synopsis>The maximum number of allowed audio streams for the endpoint</synopsis>
|
|
<description><para>
|
|
This option enforces a limit on the maximum simultaneous negotiated audio
|
|
streams allowed for the endpoint.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="max_video_streams" default="1">
|
|
<synopsis>The maximum number of allowed video streams for the endpoint</synopsis>
|
|
<description><para>
|
|
This option enforces a limit on the maximum simultaneous negotiated video
|
|
streams allowed for the endpoint.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="bundle" default="no">
|
|
<synopsis>Enable RTP bundling</synopsis>
|
|
<description><para>
|
|
With this option enabled, Asterisk will attempt to negotiate the use of bundle.
|
|
If negotiated this will result in multiple RTP streams being carried over the same
|
|
underlying transport. Note that enabling bundle will also enable the rtcp_mux option.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="webrtc" default="no">
|
|
<synopsis>Defaults and enables some options that are relevant to WebRTC</synopsis>
|
|
<description><para>
|
|
When set to "yes" this also enables the following values that are needed in
|
|
order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and
|
|
use_received_transport. The following configuration settings also get defaulted
|
|
as follows:</para>
|
|
<para>media_encryption=dtls</para>
|
|
<para>dtls_auto_generate_cert=yes (if dtls_cert_file is not set)</para>
|
|
<para>dtls_verify=fingerprint</para>
|
|
<para>dtls_setup=actpass</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>
|
|
<configOption name="follow_early_media_fork">
|
|
<synopsis>Follow SDP forked media when To tag is different</synopsis>
|
|
<description><para>
|
|
On outgoing calls, if the UAS responds with different SDP attributes
|
|
on subsequent 18X or 2XX responses (such as a port update) AND the
|
|
To tag on the subsequent response is different than that on the previous
|
|
one, follow it. This usually happens when the INVITE is forked to multiple
|
|
UASs and more than one sends an SDP answer.
|
|
</para>
|
|
<note><para>
|
|
This option must also be enabled in the <literal>system</literal>
|
|
section for it to take effect here.
|
|
</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="accept_multiple_sdp_answers" default="no">
|
|
<synopsis>Accept multiple SDP answers on non-100rel responses</synopsis>
|
|
<description><para>
|
|
On outgoing calls, if the UAS responds with different SDP attributes
|
|
on non-100rel 18X or 2XX responses (such as a port update) AND the
|
|
To tag on the subsequent response is the same as that on the previous one,
|
|
process the updated SDP. This can happen when the UAS needs to change ports
|
|
for some reason such as using a separate port for custom ringback.
|
|
</para>
|
|
<note><para>
|
|
This option must also be enabled in the <literal>system</literal>
|
|
section for it to take effect here.
|
|
</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="suppress_q850_reason_headers" default="no">
|
|
<synopsis>Suppress Q.850 Reason headers for this endpoint</synopsis>
|
|
<description><para>
|
|
Some devices can't accept multiple Reason headers and get confused
|
|
when both 'SIP' and 'Q.850' Reason headers are received. This
|
|
option allows the 'Q.850' Reason header to be suppressed.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="ignore_183_without_sdp" default="no">
|
|
<synopsis>Do not forward 183 when it doesn't contain SDP</synopsis>
|
|
<description><para>
|
|
Certain SS7 internetworking scenarios can result in a 183
|
|
to be generated for reasons other than early media. Forwarding
|
|
this 183 can cause loss of ringback tone. This flag emulates
|
|
the behavior of chan_sip and prevents these 183 responses from
|
|
being forwarded.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="stir_shaken" default="no">
|
|
<synopsis>Enable STIR/SHAKEN support on this endpoint</synopsis>
|
|
<description><para>
|
|
Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs,
|
|
the Identity header will be checked for validity. On outgoing
|
|
INVITEs, an Identity header will be added.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="stir_shaken_profile" default="">
|
|
<synopsis>STIR/SHAKEN profile containing additional configuration options</synopsis>
|
|
<description><para>
|
|
A STIR/SHAKEN profile that is defined in stir_shaken.conf. Contains
|
|
several options and rules used for STIR/SHAKEN.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="allow_unauthenticated_options" default="no">
|
|
<synopsis>Skip authentication when receiving OPTIONS requests</synopsis>
|
|
<description><para>
|
|
RFC 3261 says that the response to an OPTIONS request MUST be the
|
|
same had the request been an INVITE. Some UAs use OPTIONS requests
|
|
like a 'ping' and the expectation is that they will return a
|
|
200 OK.</para>
|
|
<para>Enabling <literal>allow_unauthenticated_options</literal>
|
|
will skip authentication of OPTIONS requests for the given
|
|
endpoint.</para>
|
|
<para>There are security implications to enabling this setting as
|
|
it can allow information disclosure to occur - specifically, if
|
|
enabled, an external party could enumerate and find the endpoint
|
|
name by sending OPTIONS requests and examining the
|
|
responses.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="security_negotiation" default="no">
|
|
<synopsis>The kind of security agreement negotiation to use. Currently, only mediasec is supported.</synopsis>
|
|
<description>
|
|
<enumlist>
|
|
<enum name="no" />
|
|
<enum name="mediasec" />
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="security_mechanisms">
|
|
<synopsis>List of security mechanisms supported.</synopsis>
|
|
<description><para>
|
|
This is a comma-delimited list of security mechanisms to use. Each security mechanism
|
|
must be in the form defined by RFC 3329 section 2.2.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="geoloc_incoming_call_profile" default="">
|
|
<synopsis>Geolocation profile to apply to incoming calls</synopsis>
|
|
<description><para>
|
|
This geolocation profile will be applied to all calls received
|
|
by the channel driver from the remote endpoint before they're
|
|
forwarded to the dialplan.
|
|
</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="geoloc_outgoing_call_profile" default="">
|
|
<synopsis>Geolocation profile to apply to outgoing calls</synopsis>
|
|
<description><para>
|
|
This geolocation profile will be applied to all calls received
|
|
by the channel driver from the dialplan before they're forwarded
|
|
the remote endpoint.
|
|
</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="send_aoc" default="no">
|
|
<synopsis>Send Advice-of-Charge messages</synopsis>
|
|
</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'. If set to <literal>google_oauth</literal> then we'll read from the
|
|
refresh_token/oauth_clientid/oauth_secret fields. The following values are valid:
|
|
</para>
|
|
<enumlist>
|
|
<enum name="md5"/>
|
|
<enum name="userpass"/>
|
|
<enum name="google_oauth"/>
|
|
</enumlist>
|
|
<para>
|
|
</para>
|
|
<note>
|
|
<para>
|
|
This setting only describes whether the password is in
|
|
plain text or has been pre-hashed with MD5. It doesn't describe
|
|
the acceptable digest algorithms we'll accept in a received
|
|
challenge.
|
|
</para>
|
|
</note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="nonce_lifetime" default="32">
|
|
<synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
|
|
</configOption>
|
|
<configOption name="md5_cred" default="">
|
|
<synopsis>MD5 Hash used for authentication.</synopsis>
|
|
<description><para>
|
|
Only used when auth_type is <literal>md5</literal>.
|
|
As an alternative to specifying a plain text password,
|
|
you can hash the username, realm and password
|
|
together one time and place the hash value here.
|
|
The input to the hash function must be in the
|
|
following format:
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
<username>:<realm>:<password>
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
For incoming authentication (asterisk is the server),
|
|
the realm must match either the realm set in this object
|
|
or the <variable>default_realm</variable> set in in the
|
|
<replaceable>global</replaceable> object.
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
For outgoing authentication (asterisk is the UAC),
|
|
the realm must match what the server will be sending
|
|
in their WWW-Authenticate header. It can't be blank
|
|
unless you expect the server to be sending a blank
|
|
realm in the header. You can't use pre-hashed
|
|
passwords with a wildcard auth object.
|
|
You can generate the hash with the following shell
|
|
command:
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
$ echo -n "myname:myrealm:mypassword" | md5sum
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
Note the '-n'. You don't want a newline to be part
|
|
of the hash.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="password">
|
|
<synopsis>Plain text password used for authentication.</synopsis>
|
|
<description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
|
|
</configOption>
|
|
<configOption name="refresh_token">
|
|
<synopsis>OAuth 2.0 refresh token</synopsis>
|
|
</configOption>
|
|
<configOption name="oauth_clientid">
|
|
<synopsis>OAuth 2.0 application's client id</synopsis>
|
|
</configOption>
|
|
<configOption name="oauth_secret">
|
|
<synopsis>OAuth 2.0 application's secret</synopsis>
|
|
</configOption>
|
|
<configOption name="realm" default="">
|
|
<synopsis>SIP realm for endpoint</synopsis>
|
|
<description><para>
|
|
For incoming authentication (asterisk is the UAS),
|
|
this is the realm to be sent on WWW-Authenticate
|
|
headers. If not specified, the <replaceable>global</replaceable>
|
|
object's <variable>default_realm</variable> will be used.
|
|
</para>
|
|
<para>
|
|
</para>
|
|
<para>
|
|
For outgoing authentication (asterisk is the UAC), this
|
|
must either be the realm the server is expected to send,
|
|
or left blank or contain a single '*' to automatically
|
|
use the realm sent by the server. If you have multiple
|
|
auth objects for an endpoint, the realm is also used to
|
|
match the auth object to the realm the server sent.
|
|
</para>
|
|
<para>
|
|
</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>
|
|
<para>
|
|
</para>
|
|
<note>
|
|
<para>
|
|
If more than one auth object with the same realm or
|
|
more than one wildcard auth object associated to
|
|
an endpoint, we can only use the first one of
|
|
each defined on the endpoint.
|
|
</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, can no longer be set, always set to 1</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, not WSS)</synopsis>
|
|
</configOption>
|
|
<configOption name="ca_list_path">
|
|
<synopsis>Path to directory containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
|
|
</configOption>
|
|
<configOption name="cert_file">
|
|
<synopsis>Certificate file for endpoint (TLS ONLY, not WSS)</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. The certificate file can be reloaded if
|
|
the filename in configuration remains unchanged.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="cipher">
|
|
<synopsis>Preferred cryptography cipher names (TLS ONLY, not WSS)</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, not WSS)</synopsis>
|
|
<description>
|
|
<para>The availability of each of these options is dependent on the
|
|
version and configuration of the underlying PJSIP library.</para>
|
|
<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="tlsv1_1" />
|
|
<enum name="tlsv1_2" />
|
|
<enum name="tlsv1_3" />
|
|
<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, not WSS)</synopsis>
|
|
<description><para>
|
|
A path to a key file can be provided. The private key file
|
|
can be reloaded if the filename in configuration remains
|
|
unchanged.
|
|
</para></description>
|
|
</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" />
|
|
<enum name="flow" />
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="require_client_cert" default="false">
|
|
<synopsis>Require client certificate (TLS ONLY, not WSS)</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, not WSS)</synopsis>
|
|
</configOption>
|
|
<configOption name="verify_server" default="false">
|
|
<synopsis>Require verification of server certificate (TLS ONLY, not WSS)</synopsis>
|
|
</configOption>
|
|
<configOption name="tos" default="false">
|
|
<synopsis>Enable TOS for the signalling sent over this transport</synopsis>
|
|
<description>
|
|
<para>See <literal>https://docs.asterisk.org/Configuration/Channel-Drivers/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://docs.asterisk.org/Configuration/Channel-Drivers/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" default="100">
|
|
<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.</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="allow_wildcard_certs" default="false">
|
|
<synopsis>Allow use of wildcards in certificates (TLS ONLY)</synopsis>
|
|
<description>
|
|
<para>In combination with verify_server, when enabled allow use of wildcards,
|
|
i.e. '*.' in certs for common,and subject alt names of type DNS for TLS
|
|
transport types. Names must start with the wildcard. Partial wildcards, e.g.
|
|
'f*.example.com' and 'foo.*.com' are not allowed. As well, names only match
|
|
against a single level meaning '*.example.com' matches 'foo.example.com',
|
|
but not 'foo.bar.example.com'.
|
|
</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="symmetric_transport" default="no">
|
|
<synopsis>Use the same transport for outgoing requests 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 respond to the OPTIONS request before the timeout,
|
|
the contact is marked unavailable. This includes time spent performing
|
|
any required DNS lookup(s) prior to sending the OPTIONS.
|
|
If <literal>0</literal> no timeout. Time in fractional seconds.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="authenticate_qualify">
|
|
<synopsis>Authenticates a qualify challenge response if needed</synopsis>
|
|
<description>
|
|
<para>If true and a qualify request receives a challenge response then
|
|
authentication is attempted before declaring the contact available.
|
|
</para>
|
|
<note><para>This option does nothing as we will always complete
|
|
the challenge response authentication if the qualify request is
|
|
challenged.
|
|
</para></note>
|
|
</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 specifying them each directly when dialing via the dialplan.
|
|
This must be used in conjunction 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_mwi_external 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>
|
|
Maximum 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> and
|
|
<replaceable>remove_unavailable</replaceable> options can help by
|
|
removing either the soonest to expire or unavailable 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="remove_unavailable" default="no">
|
|
<synopsis>Determines whether new contacts should replace unavailable ones.</synopsis>
|
|
<description><para>
|
|
The effect of this setting depends on the setting of
|
|
<replaceable>remove_existing</replaceable>.</para>
|
|
<para>If <replaceable>remove_existing</replaceable> is set to
|
|
<literal>no</literal> (default), setting remove_unavailable to
|
|
<literal>yes</literal> will remove only unavailable contacts that exceed
|
|
<replaceable>max_contacts</replaceable> to allow an incoming
|
|
REGISTER to complete sucessfully.</para>
|
|
<para>If <replaceable>remove_existing</replaceable> is set to
|
|
<literal>yes</literal>, setting remove_unavailable to
|
|
<literal>yes</literal> will prioritize unavailable contacts for removal
|
|
instead of just removing the contact that expires the soonest.</para>
|
|
<note><para>See <replaceable>remove_existing</replaceable> and
|
|
<replaceable>max_contacts</replaceable> for further information about how
|
|
these 3 settings interact.
|
|
</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 respond to the OPTIONS request before the timeout,
|
|
the contact is marked unavailable. This includes time spent performing
|
|
any required DNS lookup(s) prior to sending the OPTIONS.
|
|
If <literal>0</literal> no timeout. Time in fractional seconds.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="authenticate_qualify">
|
|
<synopsis>Authenticates a qualify challenge response if needed</synopsis>
|
|
<description>
|
|
<para>If true and a qualify request receives a challenge response then
|
|
authentication is attempted before declaring the contact available.
|
|
</para>
|
|
<note><para>This option does nothing as we will always complete
|
|
the challenge response authentication if the qualify request is
|
|
challenged.
|
|
</para></note>
|
|
</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="follow_early_media_fork">
|
|
<synopsis>Follow SDP forked media when To tag is different</synopsis>
|
|
<description><para>
|
|
On outgoing calls, if the UAS responds with different SDP attributes
|
|
on subsequent 18X or 2XX responses (such as a port update) AND the
|
|
To tag on the subsequent response is different than that on the previous
|
|
one, follow it.
|
|
</para>
|
|
<note><para>
|
|
This option must also be enabled on endpoints that require
|
|
this functionality.
|
|
</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="accept_multiple_sdp_answers">
|
|
<synopsis>Follow SDP forked media when To tag is the same</synopsis>
|
|
<description><para>
|
|
On outgoing calls, if the UAS responds with different SDP attributes
|
|
on non-100rel 18X or 2XX responses (such as a port update) AND the
|
|
To tag on the subsequent response is the same as that on the previous one,
|
|
process the updated SDP.
|
|
</para>
|
|
<note><para>
|
|
This option must also be enabled on endpoints that require
|
|
this functionality.
|
|
</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="disable_rport" default="no">
|
|
<synopsis>Disable the use of rport in outgoing requests.</synopsis>
|
|
<description><para>
|
|
Remove "rport" parameter from the outgoing requests.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type 'system' UNLESS the object name is '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="90">
|
|
<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 performance by reducing the number of database requests.
|
|
</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' UNLESS the object name is 'global'.</synopsis>
|
|
</configOption>
|
|
<configOption name="user_agent" default="Asterisk <Asterisk Version>">
|
|
<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">
|
|
<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 compatibility 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>
|
|
<configOption name="use_callerid_contact" default="no">
|
|
<synopsis>Place caller-id information into Contact header</synopsis>
|
|
<description><para>
|
|
This option will cause Asterisk to place caller-id information into
|
|
generated Contact headers.</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="send_contact_status_on_update_registration" default="no">
|
|
<synopsis>Enable sending AMI ContactStatus event when a device refreshes its registration.</synopsis>
|
|
</configOption>
|
|
<configOption name="taskprocessor_overload_trigger">
|
|
<synopsis>Trigger scope for taskprocessor overloads</synopsis>
|
|
<description><para>
|
|
This option specifies the trigger the distributor will use for
|
|
detecting taskprocessor overloads. When it detects an overload condition,
|
|
the distrubutor will stop accepting new requests until the overload is
|
|
cleared.
|
|
</para>
|
|
<enumlist>
|
|
<enum name="global"><para>(default) Any taskprocessor overload will trigger.</para></enum>
|
|
<enum name="pjsip_only"><para>Only pjsip taskprocessor overloads will trigger.</para></enum>
|
|
<enum name="none"><para>No overload detection will be performed.</para></enum>
|
|
</enumlist>
|
|
<warning><para>
|
|
The "none" and "pjsip_only" options should be used
|
|
with extreme caution and only to mitigate specific issues.
|
|
Under certain conditions they could make things worse.
|
|
</para></warning>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="norefersub" default="yes">
|
|
<synopsis>Advertise support for RFC4488 REFER subscription suppression</synopsis>
|
|
</configOption>
|
|
<configOption name="allow_sending_180_after_183" default="no">
|
|
<synopsis>Allow 180 after 183</synopsis>
|
|
<description><para>
|
|
Allow Asterisk to send 180 Ringing to an endpoint
|
|
after 183 Session Progress has been send.
|
|
If disabled Asterisk will instead send only a
|
|
183 Session Progress to the endpoint.
|
|
(default: "no")
|
|
</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="all_codecs_on_empty_reinvite" default="no">
|
|
<synopsis>If we should return all codecs on re-INVITE without SDP</synopsis>
|
|
<description><para>
|
|
On reception of a re-INVITE without SDP Asterisk will send an SDP
|
|
offer in the 200 OK response containing all configured codecs on the
|
|
endpoint, instead of simply those that have already been negotiated.
|
|
RFC 3261 specifies this as a SHOULD requirement.
|
|
</para></description>
|
|
</configOption>
|
|
</configObject>
|
|
</configFile>
|
|
</configInfo>
|
|
</docs>
|