3621 lines
148 KiB
C
3621 lines
148 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#include "asterisk.h"
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#include <pjsip.h>
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/* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
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#include <pjsip_simple.h>
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#include <pjlib.h>
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#include "asterisk/res_pjsip.h"
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#include "res_pjsip/include/res_pjsip_private.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/logger.h"
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#include "asterisk/lock.h"
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#include "asterisk/utils.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/module.h"
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#include "asterisk/threadpool.h"
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#include "asterisk/taskprocessor.h"
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#include "asterisk/uuid.h"
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#include "asterisk/sorcery.h"
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#include "asterisk/file.h"
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#include "asterisk/cli.h"
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#include "asterisk/res_pjsip_cli.h"
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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_sorcery_config</depend>
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<depend>res_sorcery_memory</depend>
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<depend>res_sorcery_astdb</depend>
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<support_level>core</support_level>
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***/
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/*** DOCUMENTATION
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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 assicated 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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<enum name="required" />
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<enum name="yes" />
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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="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 <literal>authentication</literal> object
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configured will allow connections without vertification.
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</para></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 on
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the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/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="connected_line_method" default="invite">
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<synopsis>Connected line method type</synopsis>
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<description>
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<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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<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="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">
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<synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
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</configOption>
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<configOption name="disallow">
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<synopsis>Media Codec(s) to disallow</synopsis>
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</configOption>
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<configOption name="dtmf_mode" default="rfc4733">
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<synopsis>DTMF mode</synopsis>
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<description>
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<para>This setting allows to choose the DTMF mode for endpoint communication.</para>
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<enumlist>
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<enum name="rfc4733">
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<para>DTMF is sent out of band of the main audio stream. This
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supercedes the older <emphasis>RFC-2833</emphasis> used within
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the older <literal>chan_sip</literal>.</para>
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</enum>
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<enum name="inband">
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<para>DTMF is sent as part of audio stream.</para>
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</enum>
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<enum name="info">
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<para>DTMF is sent as SIP INFO packets.</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="media_address">
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<synopsis>IP address used in SDP for media handling</synopsis>
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<description><para>
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At the time of SDP creation, the IP address defined here will be used as
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the media address for individual streams in the SDP.
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</para>
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<note><para>
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Be aware that the <literal>external_media_address</literal> option, set in Transport
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configuration, can also affect the final media address used in the SDP.
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</para></note>
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</description>
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</configOption>
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<configOption name="force_rport" default="yes">
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<synopsis>Force use of return port</synopsis>
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</configOption>
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<configOption name="ice_support" default="no">
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<synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
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</configOption>
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<configOption name="identify_by" default="username,location">
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<synopsis>Way(s) for Endpoint to be identified</synopsis>
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<description><para>
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An endpoint can be identified in multiple ways. Currently, the only supported
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option is <literal>username</literal>, which matches the endpoint based on the
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username in the From header.
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</para>
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<note><para>Endpoints can also be identified by IP address; however, that method
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of identification is not handled by this configuration option. See the documentation
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for the <literal>identify</literal> configuration section for more details on that
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method of endpoint identification. If this option is set to <literal>username</literal>
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and an <literal>identify</literal> configuration section exists for the endpoint, then
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the endpoint can be identified in multiple ways.</para></note>
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<enumlist>
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<enum name="username" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="redirect_method">
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<synopsis>How redirects received from an endpoint are handled</synopsis>
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<description><para>
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When a redirect is received from an endpoint there are multiple ways it can be handled.
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If this option is set to <literal>user</literal> the user portion of the redirect target
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is treated as an extension within the dialplan and dialed using a Local channel. If this option
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is set to <literal>uri_core</literal> the target URI is returned to the dialing application
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which dials it using the PJSIP channel driver and endpoint originally used. If this option is
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set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
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to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
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and also supporting multiple potential redirect targets. The con is that since redirection occurs
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within chan_pjsip redirecting information is not forwarded and redirection can not be
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prevented.
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</para>
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<enumlist>
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<enum name="user" />
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<enum name="uri_core" />
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<enum name="uri_pjsip" />
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</enumlist>
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</description>
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</configOption>
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<configOption name="mailboxes">
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<synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
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<description><para>
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Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
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changes happen for any of the specified mailboxes. More than one mailbox can be
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specified with a comma-delimited string. app_voicemail mailboxes must be specified
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as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
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external sources, such as through the res_external_mwi module, you must specify
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strings supported by the external system.
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</para><para>
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For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
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configuration.
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</para></description>
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</configOption>
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<configOption name="moh_suggest" default="default">
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<synopsis>Default Music On Hold class</synopsis>
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</configOption>
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<configOption name="outbound_auth">
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<synopsis>Authentication object used for outbound requests</synopsis>
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</configOption>
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<configOption name="outbound_proxy">
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<synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
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</configOption>
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<configOption name="rewrite_contact">
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<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
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<description><para>
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On inbound SIP messages from this endpoint, the Contact header will be changed to have the
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source IP address and port. This option does not affect outbound messages send to this
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endpoint.
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</para></description>
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</configOption>
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<configOption name="rtp_ipv6" default="no">
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<synopsis>Allow use of IPv6 for RTP traffic</synopsis>
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</configOption>
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<configOption name="rtp_symmetric" default="no">
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<synopsis>Enforce that RTP must be symmetric</synopsis>
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</configOption>
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<configOption name="send_diversion" default="yes">
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<synopsis>Send the Diversion header, conveying the diversion
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information to the called user agent</synopsis>
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</configOption>
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<configOption name="send_pai" default="no">
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<synopsis>Send the P-Asserted-Identity header</synopsis>
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</configOption>
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<configOption name="send_rpid" default="no">
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<synopsis>Send the Remote-Party-ID header</synopsis>
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</configOption>
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<configOption name="rpid_immediate" default="no">
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<synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
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<description>
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<para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
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or <emphasis>183 Progress</emphasis> response messages to the
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caller if the connected line information is updated before
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the call is answered. This can send a <emphasis>180 Ringing</emphasis>
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response before the call has even reached the far end. The
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caller can start hearing ringback before the far end even gets
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the call. Many phones tend to grab the first connected line
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information and refuse to update the display if it changes. The
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first information is not likely to be correct if the call
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goes to an endpoint not under the control of this Asterisk
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box.</para>
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<para>When disabled, a connected line update must wait for
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another reason to send a message with the connected line
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information to the caller before the call is answered. You can
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trigger the sending of the information by using an appropriate
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dialplan application such as <emphasis>Ringing</emphasis>.</para>
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</description>
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</configOption>
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<configOption name="timers_min_se" default="90">
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<synopsis>Minimum session timers expiration period</synopsis>
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<description><para>
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Minimium session timer expiration period. Time in seconds.
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</para></description>
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</configOption>
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<configOption name="timers" default="yes">
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<synopsis>Session timers for SIP packets</synopsis>
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<description>
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<enumlist>
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<enum name="no" />
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<enum name="yes" />
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<enum name="required" />
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<enum name="always" />
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<enum name="forced"><para>Alias of always</para></enum>
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</enumlist>
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</description>
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</configOption>
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<configOption name="timers_sess_expires" default="1800">
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<synopsis>Maximum session timer expiration period</synopsis>
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<description><para>
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Maximium session timer expiration period. Time in seconds.
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</para></description>
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</configOption>
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<configOption name="transport">
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<synopsis>Desired transport configuration</synopsis>
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<description><para>
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This will set the desired transport configuration to send SIP data through.
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</para>
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<warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
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to the first configured transport in <filename>pjsip.conf</filename> which is
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valid for the URI we are trying to contact.
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</para></warning>
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<warning><para>Transport configuration is not affected by reloads. In order to
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change transports, a full Asterisk restart is required</para></warning>
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</description>
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</configOption>
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<configOption name="trust_id_inbound" default="no">
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<synopsis>Accept identification information received from this endpoint</synopsis>
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<description><para>This option determines whether Asterisk will accept
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identification from the endpoint from headers such as P-Asserted-Identity
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or Remote-Party-ID header. This option applies both to calls originating from the
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endpoint and calls originating from Asterisk. If <literal>no</literal>, the
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configured Caller-ID from pjsip.conf will always be used as the identity for
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the endpoint.</para></description>
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</configOption>
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<configOption name="trust_id_outbound" default="no">
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<synopsis>Send private identification details to the endpoint.</synopsis>
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<description><para>This option determines whether res_pjsip will send private
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identification information to the endpoint. If <literal>no</literal>,
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private Caller-ID information will not be forwarded to the endpoint.
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"Private" in this case refers to any method of restricting identification.
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Example: setting <replaceable>callerid_privacy</replaceable> to any
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<literal>prohib</literal> variation.
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Example: If <replaceable>trust_id_inbound</replaceable> is set to
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<literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
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header in a SIP request or response would indicate the identification
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provided in the request is private.</para></description>
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</configOption>
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<configOption name="type">
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<synopsis>Must be of type 'endpoint'.</synopsis>
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</configOption>
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<configOption name="use_ptime" default="no">
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<synopsis>Use Endpoint's requested packetisation interval</synopsis>
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</configOption>
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<configOption name="use_avpf" default="no">
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<synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
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endpoint.</synopsis>
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<description><para>
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If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
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profile for all media offers on outbound calls and media updates and will
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decline media offers not using the AVPF or SAVPF profile.
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</para><para>
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If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
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profile for all media offers on outbound calls and media updates, and will
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decline media offers not using the AVP or SAVP profile.
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</para></description>
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</configOption>
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<configOption name="force_avp" default="no">
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<synopsis>Determines whether res_pjsip will use and enforce usage of AVP,
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regardless of the RTP profile in use for this endpoint.</synopsis>
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<description><para>
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If set to <literal>yes</literal>, res_pjsip will use the AVP, AVPF, SAVP, or
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SAVPF RTP profile for all media offers on outbound calls and media updates including
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those for DTLS-SRTP streams.
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</para><para>
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If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
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depending on configuration.
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</para></description>
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</configOption>
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<configOption name="media_use_received_transport" default="no">
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<synopsis>Determines whether res_pjsip will use the media transport received in the
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offer SDP in the corresponding answer SDP.</synopsis>
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<description><para>
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If set to <literal>yes</literal>, res_pjsip will use the received media transport.
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</para><para>
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If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
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depending on configuration.
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</para></description>
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</configOption>
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|
<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="inband_progress" default="no">
|
|
<synopsis>Determines whether chan_pjsip will indicate ringing using inband
|
|
progress.</synopsis>
|
|
<description><para>
|
|
If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
|
|
when told to indicate ringing and will immediately start sending ringing
|
|
as audio.
|
|
</para><para>
|
|
If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
|
|
to indicate ringing and will NOT send it as audio.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="call_group">
|
|
<synopsis>The numeric pickup groups for a channel.</synopsis>
|
|
<description><para>
|
|
Can be set to a comma separated list of numbers or ranges between the values
|
|
of 0-63 (maximum of 64 groups).
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="pickup_group">
|
|
<synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
|
|
<description><para>
|
|
Can be set to a comma separated list of numbers or ranges between the values
|
|
of 0-63 (maximum of 64 groups).
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="named_call_group">
|
|
<synopsis>The named pickup groups for a channel.</synopsis>
|
|
<description><para>
|
|
Can be set to a comma separated list of case sensitive strings limited by
|
|
supported line length.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="named_pickup_group">
|
|
<synopsis>The named pickup groups that a channel can pickup.</synopsis>
|
|
<description><para>
|
|
Can be set to a comma separated list of case sensitive strings limited by
|
|
supported line length.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="device_state_busy_at" default="0">
|
|
<synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
|
|
<description><para>
|
|
When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
|
|
PJSIP channel driver will return busy as the device state instead of in use.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="t38_udptl" default="no">
|
|
<synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
|
|
<description><para>
|
|
If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
|
|
and relayed.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="t38_udptl_ec" default="none">
|
|
<synopsis>T.38 UDPTL error correction method</synopsis>
|
|
<description>
|
|
<enumlist>
|
|
<enum name="none"><para>
|
|
No error correction should be used.
|
|
</para></enum>
|
|
<enum name="fec"><para>
|
|
Forward error correction should be used.
|
|
</para></enum>
|
|
<enum name="redundancy"><para>
|
|
Redundacy error correction should be used.
|
|
</para></enum>
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="t38_udptl_maxdatagram" default="0">
|
|
<synopsis>T.38 UDPTL maximum datagram size</synopsis>
|
|
<description><para>
|
|
This option can be set to override the maximum datagram of a remote endpoint for broken
|
|
endpoints.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="fax_detect" default="no">
|
|
<synopsis>Whether CNG tone detection is enabled</synopsis>
|
|
<description><para>
|
|
This option can be set to send the session to the fax extension when a CNG tone is
|
|
detected.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="t38_udptl_nat" default="no">
|
|
<synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
|
|
<description><para>
|
|
When enabled the UDPTL stack will send UDPTL packets to the source address of
|
|
received packets.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="t38_udptl_ipv6" default="no">
|
|
<synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
|
|
<description><para>
|
|
When enabled the UDPTL stack will use IPv6.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="tone_zone">
|
|
<synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="language">
|
|
<synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="one_touch_recording" default="no">
|
|
<synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
|
|
<see-also>
|
|
<ref type="configOption">record_on_feature</ref>
|
|
<ref type="configOption">record_off_feature</ref>
|
|
</see-also>
|
|
</configOption>
|
|
<configOption name="record_on_feature" default="automixmon">
|
|
<synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
|
|
<description>
|
|
<para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
|
|
feature will be enabled for the channel. The feature designated here can be any built-in
|
|
or dynamic feature defined in features.conf.</para>
|
|
<note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
|
|
</description>
|
|
<see-also>
|
|
<ref type="configOption">one_touch_recording</ref>
|
|
<ref type="configOption">record_off_feature</ref>
|
|
</see-also>
|
|
</configOption>
|
|
<configOption name="record_off_feature" default="automixmon">
|
|
<synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
|
|
<description>
|
|
<para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
|
|
feature will be enabled for the channel. The feature designated here can be any built-in
|
|
or dynamic feature defined in features.conf.</para>
|
|
<note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
|
|
</description>
|
|
<see-also>
|
|
<ref type="configOption">one_touch_recording</ref>
|
|
<ref type="configOption">record_on_feature</ref>
|
|
</see-also>
|
|
</configOption>
|
|
<configOption name="rtp_engine" default="asterisk">
|
|
<synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
|
|
</configOption>
|
|
<configOption name="allow_transfer" default="yes">
|
|
<synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
|
|
</configOption>
|
|
<configOption name="user_eq_phone" default="no">
|
|
<synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
|
|
</configOption>
|
|
<configOption name="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://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="tos_video">
|
|
<synopsis>DSCP TOS bits for video streams</synopsis>
|
|
<description><para>
|
|
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="cos_audio">
|
|
<synopsis>Priority for audio streams</synopsis>
|
|
<description><para>
|
|
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="cos_video">
|
|
<synopsis>Priority for video streams</synopsis>
|
|
<description><para>
|
|
See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="allow_subscribe" default="yes">
|
|
<synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
|
|
</configOption>
|
|
<configOption name="sub_min_expiry" default="60">
|
|
<synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="from_user">
|
|
<synopsis>Username to use in From header for requests to this endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="mwi_from_user">
|
|
<synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="from_domain">
|
|
<synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
|
|
</configOption>
|
|
<configOption name="dtls_verify">
|
|
<synopsis>Verify that the provided peer certificate is valid</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_rekey">
|
|
<synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para><para>
|
|
If this is not set or the value provided is 0 rekeying will be disabled.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_cert_file">
|
|
<synopsis>Path to certificate file to present to peer</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_private_key">
|
|
<synopsis>Path to private key for certificate file</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_cipher">
|
|
<synopsis>Cipher to use for DTLS negotiation</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para>
|
|
<para>Many options for acceptable ciphers. See link for more:</para>
|
|
<para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_ca_file">
|
|
<synopsis>Path to certificate authority certificate</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_ca_path">
|
|
<synopsis>Path to a directory containing certificate authority certificates</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="dtls_setup">
|
|
<synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
|
|
<description>
|
|
<para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para>
|
|
<enumlist>
|
|
<enum name="active"><para>
|
|
res_pjsip will make a connection to the peer.
|
|
</para></enum>
|
|
<enum name="passive"><para>
|
|
res_pjsip will accept connections from the peer.
|
|
</para></enum>
|
|
<enum name="actpass"><para>
|
|
res_pjsip will offer and accept connections from the peer.
|
|
</para></enum>
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="dtls_fingerprint">
|
|
<synopsis>Type of hash to use for the DTLS fingerprint in the SDP.</synopsis>
|
|
<description>
|
|
<para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>dtls</literal>.
|
|
</para>
|
|
<enumlist>
|
|
<enum name="SHA-256"></enum>
|
|
<enum name="SHA-1"></enum>
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="srtp_tag_32">
|
|
<synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
|
|
<description><para>
|
|
This option only applies if <replaceable>media_encryption</replaceable> is
|
|
set to <literal>sdes</literal> or <literal>dtls</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="set_var">
|
|
<synopsis>Variable set on a channel involving the endpoint.</synopsis>
|
|
<description><para>
|
|
When a new channel is created using the endpoint set the specified
|
|
variable(s) on that channel. For multiple channel variables specify
|
|
multiple 'set_var'(s).
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="message_context">
|
|
<synopsis>Context to route incoming MESSAGE requests to.</synopsis>
|
|
<description><para>
|
|
If specified, incoming MESSAGE requests will be routed to the indicated
|
|
dialplan context. If no <replaceable>message_context</replaceable> is
|
|
specified, then the <replaceable>context</replaceable> setting is used.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="accountcode">
|
|
<synopsis>An accountcode to set automatically on any channels created for this endpoint.</synopsis>
|
|
<description><para>
|
|
If specified, any channel created for this endpoint will automatically
|
|
have this accountcode set on it.
|
|
</para></description>
|
|
</configOption>
|
|
</configObject>
|
|
<configObject name="auth">
|
|
<synopsis>Authentication type</synopsis>
|
|
<description><para>
|
|
Authentication objects hold the authentication information for use
|
|
by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
|
|
This also allows for multiple objects to use a single auth object. See
|
|
the <literal>auth_type</literal> config option for password style choices.
|
|
</para></description>
|
|
<configOption name="auth_type" default="userpass">
|
|
<synopsis>Authentication type</synopsis>
|
|
<description><para>
|
|
This option specifies which of the password style config options should be read
|
|
when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
|
|
then we'll read from the 'password' option. For <literal>md5</literal> we'll read
|
|
from 'md5_cred'.
|
|
</para>
|
|
<enumlist>
|
|
<enum name="md5"/>
|
|
<enum name="userpass"/>
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="nonce_lifetime" default="32">
|
|
<synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
|
|
</configOption>
|
|
<configOption name="md5_cred">
|
|
<synopsis>MD5 Hash used for authentication.</synopsis>
|
|
<description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
|
|
</configOption>
|
|
<configOption name="password">
|
|
<synopsis>PlainText password used for authentication.</synopsis>
|
|
<description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
|
|
</configOption>
|
|
<configOption name="realm" default="asterisk">
|
|
<synopsis>SIP realm for endpoint</synopsis>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be 'auth'</synopsis>
|
|
</configOption>
|
|
<configOption name="username">
|
|
<synopsis>Username to use for account</synopsis>
|
|
</configOption>
|
|
</configObject>
|
|
<configObject name="domain_alias">
|
|
<synopsis>Domain Alias</synopsis>
|
|
<description><para>
|
|
Signifies that a domain is an alias. If the domain on a session is
|
|
not found to match an AoR then this object is used to see if we have
|
|
an alias for the AoR to which the endpoint is binding. This objects
|
|
name as defined in configuration should be the domain alias and a
|
|
config option is provided to specify the domain to be aliased.
|
|
</para></description>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type 'domain_alias'.</synopsis>
|
|
</configOption>
|
|
<configOption name="domain">
|
|
<synopsis>Domain to be aliased</synopsis>
|
|
</configOption>
|
|
</configObject>
|
|
<configObject name="transport">
|
|
<synopsis>SIP Transport</synopsis>
|
|
<description><para>
|
|
<emphasis>Transports</emphasis>
|
|
</para>
|
|
<para>There are different transports and protocol derivatives
|
|
supported by <literal>res_pjsip</literal>. They are in order of
|
|
preference: UDP, TCP, and WebSocket (WS).</para>
|
|
<note><para>Changes to transport configuration in pjsip.conf will only be
|
|
effected on a complete restart of Asterisk. A module reload
|
|
will not suffice.</para></note>
|
|
</description>
|
|
<configOption name="async_operations" default="1">
|
|
<synopsis>Number of simultaneous Asynchronous Operations</synopsis>
|
|
</configOption>
|
|
<configOption name="bind">
|
|
<synopsis>IP Address and optional port to bind to for this transport</synopsis>
|
|
</configOption>
|
|
<configOption name="ca_list_file">
|
|
<synopsis>File containing a list of certificates to read (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="ca_list_path">
|
|
<synopsis>Path to directory containing a list of certificates to read (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="cert_file">
|
|
<synopsis>Certificate file for endpoint (TLS ONLY)</synopsis>
|
|
<description><para>
|
|
A path to a .crt or .pem file can be provided. However, only
|
|
the certificate is read from the file, not the private key.
|
|
The <literal>priv_key_file</literal> option must supply a
|
|
matching key file.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="cipher">
|
|
<synopsis>Preferred cryptography cipher names (TLS ONLY)</synopsis>
|
|
<description>
|
|
<para>Comma separated list of cipher names or numeric equivalents.
|
|
Numeric equivalents can be either decimal or hexadecimal (0xX).
|
|
</para>
|
|
<para>There are many cipher names. Use the CLI command
|
|
<literal>pjsip list ciphers</literal> to see a list of cipher
|
|
names available for your installation. See link for more:</para>
|
|
<para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES
|
|
</para>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="domain">
|
|
<synopsis>Domain the transport comes from</synopsis>
|
|
</configOption>
|
|
<configOption name="external_media_address">
|
|
<synopsis>External IP address to use in RTP handling</synopsis>
|
|
<description><para>
|
|
When a request or response is sent out, if the destination of the
|
|
message is outside the IP network defined in the option <literal>localnet</literal>,
|
|
and the media address in the SDP is within the localnet network, then the
|
|
media address in the SDP will be rewritten to the value defined for
|
|
<literal>external_media_address</literal>.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="external_signaling_address">
|
|
<synopsis>External address for SIP signalling</synopsis>
|
|
</configOption>
|
|
<configOption name="external_signaling_port" default="0">
|
|
<synopsis>External port for SIP signalling</synopsis>
|
|
</configOption>
|
|
<configOption name="method">
|
|
<synopsis>Method of SSL transport (TLS ONLY)</synopsis>
|
|
<description>
|
|
<enumlist>
|
|
<enum name="default">
|
|
<para>The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.</para>
|
|
</enum>
|
|
<enum name="unspecified">
|
|
<para>This option is equivalent to setting 'default'</para>
|
|
</enum>
|
|
<enum name="tlsv1" />
|
|
<enum name="sslv2" />
|
|
<enum name="sslv3" />
|
|
<enum name="sslv23" />
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="local_net">
|
|
<synopsis>Network to consider local (used for NAT purposes).</synopsis>
|
|
<description><para>This must be in CIDR or dotted decimal format with the IP
|
|
and mask separated with a slash ('/').</para></description>
|
|
</configOption>
|
|
<configOption name="password">
|
|
<synopsis>Password required for transport</synopsis>
|
|
</configOption>
|
|
<configOption name="priv_key_file">
|
|
<synopsis>Private key file (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="protocol" default="udp">
|
|
<synopsis>Protocol to use for SIP traffic</synopsis>
|
|
<description>
|
|
<enumlist>
|
|
<enum name="udp" />
|
|
<enum name="tcp" />
|
|
<enum name="tls" />
|
|
<enum name="ws" />
|
|
<enum name="wss" />
|
|
</enumlist>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="require_client_cert" default="false">
|
|
<synopsis>Require client certificate (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type 'transport'.</synopsis>
|
|
</configOption>
|
|
<configOption name="verify_client" default="false">
|
|
<synopsis>Require verification of client certificate (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="verify_server" default="false">
|
|
<synopsis>Require verification of server certificate (TLS ONLY)</synopsis>
|
|
</configOption>
|
|
<configOption name="tos" default="false">
|
|
<synopsis>Enable TOS for the signalling sent over this transport</synopsis>
|
|
<description>
|
|
<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
|
|
for more information on this parameter.</para>
|
|
<note><para>This option does not apply to the <replaceable>ws</replaceable>
|
|
or the <replaceable>wss</replaceable> protocols.</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="cos" default="false">
|
|
<synopsis>Enable COS for the signalling sent over this transport</synopsis>
|
|
<description>
|
|
<para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
|
|
for more information on this parameter.</para>
|
|
<note><para>This option does not apply to the <replaceable>ws</replaceable>
|
|
or the <replaceable>wss</replaceable> protocols.</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="websocket_write_timeout">
|
|
<synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
|
|
<description>
|
|
<para>If a websocket connection accepts input slowly, the timeout
|
|
for writes to it can be increased to keep it from being disconnected.
|
|
Value is in milliseconds; default is 100 ms.</para>
|
|
</description>
|
|
</configOption>
|
|
</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="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>
|
|
</configObject>
|
|
<configObject name="aor">
|
|
<synopsis>The configuration for a location of an endpoint</synopsis>
|
|
<description><para>
|
|
An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
|
|
AoRs are specified, an endpoint will not be reachable by Asterisk.
|
|
Beyond that, an AoR has other uses within Asterisk, such as inbound
|
|
registration.
|
|
</para><para>
|
|
An <literal>AoR</literal> is a way to allow dialing a group
|
|
of <literal>Contacts</literal> that all use the same
|
|
<literal>endpoint</literal> for calls.
|
|
</para><para>
|
|
This can be used as another way of grouping a list of contacts to dial
|
|
rather than specifing them each directly when dialing via the dialplan.
|
|
This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
|
|
</para><para>
|
|
Registrations: For Asterisk to match an inbound registration to an endpoint,
|
|
the AoR object name must match the user portion of the SIP URI in the "To:"
|
|
header of the inbound SIP registration. That will usually be equivalent
|
|
to the "user name" set in your hard or soft phones configuration.
|
|
</para></description>
|
|
<configOption name="contact">
|
|
<synopsis>Permanent contacts assigned to AoR</synopsis>
|
|
<description><para>
|
|
Contacts specified will be called whenever referenced
|
|
by <literal>chan_pjsip</literal>.
|
|
</para><para>
|
|
Use a separate "contact=" entry for each contact required. Contacts
|
|
are specified using a SIP URI.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="default_expiration" default="3600">
|
|
<synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
|
|
</configOption>
|
|
<configOption name="mailboxes">
|
|
<synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
|
|
<description><para>This option applies when an external entity subscribes to an AoR
|
|
for Message Waiting Indications. The mailboxes specified will be subscribed to.
|
|
More than one mailbox can be specified with a comma-delimited string.
|
|
app_voicemail mailboxes must be specified as mailbox@context;
|
|
for example: mailboxes=6001@default. For mailboxes provided by external sources,
|
|
such as through the res_external_mwi module, you must specify strings supported by
|
|
the external system.
|
|
</para><para>
|
|
For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
|
|
endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="maximum_expiration" default="7200">
|
|
<synopsis>Maximum time to keep an AoR</synopsis>
|
|
<description><para>
|
|
Maximium time to keep a peer with explicit expiration. Time in seconds.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="max_contacts" default="0">
|
|
<synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
|
|
<description><para>
|
|
Maximum number of contacts that can associate with this AoR. This value does
|
|
not affect the number of contacts that can be added with the "contact" option.
|
|
It only limits contacts added through external interaction, such as
|
|
registration.
|
|
</para>
|
|
<note><para>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 explict 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
|
|
the existing contact that was registered against it?
|
|
</para>
|
|
<note><para>This should be set to <literal>yes</literal> and
|
|
<replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
|
|
wish to stick with the older <literal>chan_sip</literal> behaviour.
|
|
</para></note>
|
|
</description>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type 'aor'.</synopsis>
|
|
</configOption>
|
|
<configOption name="qualify_frequency" default="0">
|
|
<synopsis>Interval at which to qualify an AoR</synopsis>
|
|
<description><para>
|
|
Interval between attempts to qualify the AoR for reachability.
|
|
If <literal>0</literal> never qualify. Time in seconds.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="authenticate_qualify" default="no">
|
|
<synopsis>Authenticates a qualify request if needed</synopsis>
|
|
<description><para>
|
|
If true and a qualify request receives a challenge or authenticate response
|
|
authentication is attempted before declaring the contact available.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="outbound_proxy">
|
|
<synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
|
|
<description><para>
|
|
If set the provided URI will be used as the outbound proxy when an
|
|
OPTIONS request is sent to a contact for qualify purposes.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="support_path">
|
|
<synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
|
|
<description><para>
|
|
When this option is enabled, the Path headers in register requests will be saved
|
|
and its contents will be used in Route headers for outbound out-of-dialog requests
|
|
and in Path headers for outbound 200 responses. Path support will also be indicated
|
|
in the Supported header.
|
|
</para></description>
|
|
</configOption>
|
|
</configObject>
|
|
<configObject name="system">
|
|
<synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
|
|
<description><para>
|
|
The settings in this section are global. In addition to being global, the values will
|
|
not be re-evaluated when a reload is performed. This is because the values must be set
|
|
before the SIP stack is initialized. The only way to reset these values is to either
|
|
restart Asterisk, or unload res_pjsip.so and then load it again.
|
|
</para></description>
|
|
<configOption name="timer_t1" default="500">
|
|
<synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
|
|
<description><para>
|
|
Timer T1 is the base for determining how long to wait before retransmitting
|
|
requests that receive no response when using an unreliable transport (e.g. UDP).
|
|
For more information on this timer, see RFC 3261, Section 17.1.1.1.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="timer_b" default="32000">
|
|
<synopsis>Set transaction timer B value (milliseconds).</synopsis>
|
|
<description><para>
|
|
Timer B determines the maximum amount of time to wait after sending an INVITE
|
|
request before terminating the transaction. It is recommended that this be set
|
|
to 64 * Timer T1, but it may be set higher if desired. For more information on
|
|
this timer, see RFC 3261, Section 17.1.1.1.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="compact_headers" default="no">
|
|
<synopsis>Use the short forms of common SIP header names.</synopsis>
|
|
</configOption>
|
|
<configOption name="threadpool_initial_size" default="0">
|
|
<synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
|
|
</configOption>
|
|
<configOption name="threadpool_auto_increment" default="5">
|
|
<synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
|
|
</configOption>
|
|
<configOption name="threadpool_idle_timeout" default="60">
|
|
<synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
|
|
</configOption>
|
|
<configOption name="threadpool_max_size" default="0">
|
|
<synopsis>Maximum number of threads in the res_pjsip threadpool.
|
|
A value of 0 indicates no maximum.</synopsis>
|
|
</configOption>
|
|
<configOption name="disable_tcp_switch" default="yes">
|
|
<synopsis>Disable automatic switching from UDP to TCP transports.</synopsis>
|
|
<description><para>
|
|
Disable automatic switching from UDP to TCP transports if outgoing
|
|
request is too large. See RFC 3261 section 18.1.1.
|
|
</para></description>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type 'system'.</synopsis>
|
|
</configOption>
|
|
</configObject>
|
|
<configObject name="global">
|
|
<synopsis>Options that apply globally to all SIP communications</synopsis>
|
|
<description><para>
|
|
The settings in this section are global. Unlike options in the <literal>system</literal>
|
|
section, these options can be refreshed by performing a reload.
|
|
</para></description>
|
|
<configOption name="max_forwards" default="70">
|
|
<synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
|
|
</configOption>
|
|
<configOption name="keep_alive_interval" default="0">
|
|
<synopsis>The interval (in seconds) to send keepalives to active connection-oriented transports.</synopsis>
|
|
</configOption>
|
|
<configOption name="type">
|
|
<synopsis>Must be of type '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="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="debug" default="no">
|
|
<synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
|
|
a host address</synopsis>
|
|
</configOption>
|
|
<configOption name="endpoint_identifier_order" default="ip,username,anonymous">
|
|
<synopsis>The order by which endpoint identifiers are processed and checked.
|
|
Identifier names are usually derived from and can be found in the endpoint
|
|
identifier module itself (res_pjsip_endpoint_identifier_*)</synopsis>
|
|
</configOption>
|
|
</configObject>
|
|
</configFile>
|
|
</configInfo>
|
|
<manager name="PJSIPQualify" language="en_US">
|
|
<synopsis>
|
|
Qualify a chan_pjsip endpoint.
|
|
</synopsis>
|
|
<syntax>
|
|
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
|
|
<parameter name="Endpoint" required="true">
|
|
<para>The endpoint you want to qualify.</para>
|
|
</parameter>
|
|
</syntax>
|
|
<description>
|
|
<para>Qualify a chan_pjsip endpoint.</para>
|
|
</description>
|
|
</manager>
|
|
<managerEvent language="en_US" name="IdentifyDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about an identify section.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'identify'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="Endpoint">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='endpoint']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Match">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="EndpointName">
|
|
<para>The name of the endpoint associated with this information.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="AorDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'aor'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="MinimumExpiration">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MaximumExpiration">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DefaultExpiration">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="QualifyFrequency">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AuthenticateQualify">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MaxContacts">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RemoveExisting">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Mailboxes">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="OutboundProxy">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SupportPath">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TotalContacts">
|
|
<para>The total number of contacts associated with this AoR.</para>
|
|
</parameter>
|
|
<parameter name="ContactsRegistered">
|
|
<para>The number of non-permanent contacts associated with this AoR.</para>
|
|
</parameter>
|
|
<parameter name="EndpointName">
|
|
<para>The name of the endpoint associated with this information.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="AuthDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about an authentication section.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'auth'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="Username">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Password">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Md5Cred">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Realm">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="NonceLifetime">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AuthType">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="EndpointName">
|
|
<para>The name of the endpoint associated with this information.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="TransportDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about an authentication section.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'transport'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="Protocol">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='protocol']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Bind">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='bind']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AsycOperations">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='async_operations']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CaListFile">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_file']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CaListPath">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_path']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CertFile">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cert_file']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="PrivKeyFile">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='priv_key_file']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Password">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='password']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ExternalSignalingAddress">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_address']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ExternalSignalingPort">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_port']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ExternalMediaAddress">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_media_address']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Domain">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='domain']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="VerifyServer">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_server']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="VerifyClient">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_client']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RequireClientCert">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='require_client_cert']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Method">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='method']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Cipher">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cipher']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="LocalNet">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='local_net']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Tos">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='tos']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Cos">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cos']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="WebsocketWriteTimeout">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='websocket_write_timeout']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="EndpointName">
|
|
<para>The name of the endpoint associated with this information.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="EndpointDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about an endpoint section.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'endpoint'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="Context">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='context']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Disallow">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disallow']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Allow">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtmfMode">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtmf_mode']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RtpIpv6">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_ipv6']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RtpSymmetric">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_symmetric']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="IceSupport">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='ice_support']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="UsePtime">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_ptime']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ForceRport">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_rport']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RewriteContact">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rewrite_contact']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Transport">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='transport']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="OutboundProxy">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MohSuggest">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_suggest']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="100rel">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='100rel']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Timers">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TimersMinSe">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_min_se']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TimersSessExpires">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_sess_expires']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Auth">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='auth']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="OutboundAuth">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_auth']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Aors">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aors']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MediaAddress">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_address']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="IdentifyBy">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='identify_by']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DirectMedia">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DirectMediaMethod">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_method']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ConnectedLineMethod">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='connected_line_method']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DirectMediaGlareMitigation">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_glare_mitigation']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DisableDirectMediaOnNat">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disable_direct_media_on_nat']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Callerid">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CalleridPrivacy">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_privacy']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CalleridTag">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_tag']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TrustIdInbound">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_inbound']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TrustIdOutbound">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_outbound']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SendPai">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_pai']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SendRpid">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_rpid']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SendDiversion">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_diversion']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Mailboxes">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mailboxes']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AggregateMwi">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aggregate_mwi']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MediaEncryption">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MediaEncryptionOptimistic">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption_optimistic']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="UseAvpf">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_avpf']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ForceAvp">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_avp']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MediaUseReceivedTransport">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_use_received_transport']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="OneTouchRecording">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='one_touch_recording']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="InbandProgress">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='inband_progress']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CallGroup">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='call_group']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="PickupGroup">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='pickup_group']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="NamedCallGroup">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_call_group']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="NamedPickupGroup">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_pickup_group']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DeviceStateBusyAt">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='device_state_busy_at']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="T38Udptl">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="T38UdptlEc">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ec']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="T38UdptlMaxdatagram">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_maxdatagram']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="FaxDetect">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='fax_detect']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="T38UdptlNat">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_nat']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="T38UdptlIpv6">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ipv6']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="ToneZone">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tone_zone']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Language">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='language']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RecordOnFeature">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_on_feature']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RecordOffFeature">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_off_feature']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AllowTransfer">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="UserEqPhone">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MohPassthrough">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SdpOwner">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SdpSession">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_session']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TosAudio">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_audio']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="TosVideo">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_video']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CosAudio">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_audio']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="CosVideo">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_video']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="AllowSubscribe">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_subscribe']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SubMinExpiry">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sub_min_expiry']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="FromUser">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_user']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="FromDomain">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_domain']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MwiFromUser">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mwi_from_user']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RtpEngine">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_engine']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsVerify">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_verify']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsRekey">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_rekey']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsCertFile">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cert_file']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsPrivateKey">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_private_key']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsCipher">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cipher']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsCaFile">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_file']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsCaPath">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_path']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DtlsSetup">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_setup']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SrtpTag32">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='srtp_tag_32']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="RedirectMethod">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='redirect_method']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="SetVar">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='set_var']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="MessageContext">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='message_context']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="Accountcode">
|
|
<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/></para>
|
|
</parameter>
|
|
<parameter name="DeviceState">
|
|
<para>The aggregate device state for this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="ActiveChannels">
|
|
<para>The number of active channels associated with this endpoint.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="ContactStatusDetail">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about a contact's status.</synopsis>
|
|
<syntax>
|
|
<parameter name="AOR">
|
|
<para>The AoR that owns this contact.</para>
|
|
</parameter>
|
|
<parameter name="URI">
|
|
<para>This contact's URI.</para>
|
|
</parameter>
|
|
<parameter name="Status">
|
|
<para>This contact's status.</para>
|
|
<enumlist>
|
|
<enum name="Reachable"/>
|
|
<enum name="Unreachable"/>
|
|
</enumlist>
|
|
</parameter>
|
|
<parameter name="RoundtripUsec">
|
|
<para>The round trip time in microseconds.</para>
|
|
</parameter>
|
|
<parameter name="EndpointName">
|
|
<para>The name of the endpoint associated with this information.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<managerEvent language="en_US" name="EndpointList">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide details about a contact's status.</synopsis>
|
|
<syntax>
|
|
<parameter name="ObjectType">
|
|
<para>The object's type. This will always be 'endpoint'.</para>
|
|
</parameter>
|
|
<parameter name="ObjectName">
|
|
<para>The name of this object.</para>
|
|
</parameter>
|
|
<parameter name="Transport">
|
|
<para>The transport configurations associated with this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="Aor">
|
|
<para>The aor configurations associated with this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="Auths">
|
|
<para>The inbound authentication configurations associated with this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="OutboundAuths">
|
|
<para>The outbound authentication configurations associated with this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="DeviceState">
|
|
<para>The aggregate device state for this endpoint.</para>
|
|
</parameter>
|
|
<parameter name="ActiveChannels">
|
|
<para>The number of active channels associated with this endpoint.</para>
|
|
</parameter>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
<manager name="PJSIPShowEndpoints" language="en_US">
|
|
<synopsis>
|
|
Lists PJSIP endpoints.
|
|
</synopsis>
|
|
<syntax />
|
|
<description>
|
|
<para>
|
|
Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
|
|
is raised that contains relevant attributes and status information. Once all
|
|
endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
|
|
</para>
|
|
</description>
|
|
<responses>
|
|
<list-elements>
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointList'])" />
|
|
</list-elements>
|
|
<managerEvent language="en_US" name="EndpointListComplete">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide final information about an endpoint list.</synopsis>
|
|
<syntax>
|
|
<parameter name="EventList"/>
|
|
<parameter name="ListItems"/>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
</responses>
|
|
</manager>
|
|
<manager name="PJSIPShowEndpoint" language="en_US">
|
|
<synopsis>
|
|
Detail listing of an endpoint and its objects.
|
|
</synopsis>
|
|
<syntax>
|
|
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
|
|
<parameter name="Endpoint" required="true">
|
|
<para>The endpoint to list.</para>
|
|
</parameter>
|
|
</syntax>
|
|
<description>
|
|
<para>
|
|
Provides a detailed listing of options for a given endpoint. Events are issued
|
|
showing the configuration and status of the endpoint and associated objects. These
|
|
events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
|
|
<literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
|
|
<literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
|
|
associated (for instance AoRs). Once all detail events have been raised a final
|
|
<literal>EndpointDetailComplete</literal> event is issued.
|
|
</para>
|
|
</description>
|
|
<responses>
|
|
<list-elements>
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointDetail'])" />
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='IdentifyDetail'])" />
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactStatusDetail'])" />
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthDetail'])" />
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='TransportDetail'])" />
|
|
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AorDetail'])" />
|
|
</list-elements>
|
|
<managerEvent language="en_US" name="EndpointDetailComplete">
|
|
<managerEventInstance class="EVENT_FLAG_COMMAND">
|
|
<synopsis>Provide final information about endpoint details.</synopsis>
|
|
<syntax>
|
|
<parameter name="EventList"/>
|
|
<parameter name="ListItems"/>
|
|
</syntax>
|
|
</managerEventInstance>
|
|
</managerEvent>
|
|
</responses>
|
|
</manager>
|
|
***/
|
|
|
|
#define MOD_DATA_CONTACT "contact"
|
|
|
|
static pjsip_endpoint *ast_pjsip_endpoint;
|
|
|
|
static struct ast_threadpool *sip_threadpool;
|
|
|
|
static int register_service_noref(void *data)
|
|
{
|
|
pjsip_module **module = data;
|
|
if (!ast_pjsip_endpoint) {
|
|
ast_log(LOG_ERROR, "There is no PJSIP endpoint. Unable to register services\n");
|
|
return -1;
|
|
}
|
|
if (pjsip_endpt_register_module(ast_pjsip_endpoint, *module) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Unable to register module %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
|
|
return -1;
|
|
}
|
|
ast_debug(1, "Registered SIP service %.*s (%p)\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name), *module);
|
|
return 0;
|
|
}
|
|
|
|
static int register_service(void *data)
|
|
{
|
|
int res;
|
|
|
|
if (!(res = register_service_noref(data))) {
|
|
ast_module_ref(ast_module_info->self);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
int internal_sip_register_service(pjsip_module *module)
|
|
{
|
|
return ast_sip_push_task_synchronous(NULL, register_service_noref, &module);
|
|
}
|
|
|
|
int ast_sip_register_service(pjsip_module *module)
|
|
{
|
|
return ast_sip_push_task_synchronous(NULL, register_service, &module);
|
|
}
|
|
|
|
static int unregister_service_noref(void *data)
|
|
{
|
|
pjsip_module **module = data;
|
|
if (!ast_pjsip_endpoint) {
|
|
return -1;
|
|
}
|
|
pjsip_endpt_unregister_module(ast_pjsip_endpoint, *module);
|
|
ast_debug(1, "Unregistered SIP service %.*s\n", (int) pj_strlen(&(*module)->name), pj_strbuf(&(*module)->name));
|
|
return 0;
|
|
}
|
|
|
|
static int unregister_service(void *data)
|
|
{
|
|
int res;
|
|
|
|
if (!(res = unregister_service_noref(data))) {
|
|
ast_module_unref(ast_module_info->self);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
int internal_sip_unregister_service(pjsip_module *module)
|
|
{
|
|
return ast_sip_push_task_synchronous(NULL, unregister_service_noref, &module);
|
|
}
|
|
|
|
void ast_sip_unregister_service(pjsip_module *module)
|
|
{
|
|
ast_sip_push_task_synchronous(NULL, unregister_service, &module);
|
|
}
|
|
|
|
static struct ast_sip_authenticator *registered_authenticator;
|
|
|
|
int ast_sip_register_authenticator(struct ast_sip_authenticator *auth)
|
|
{
|
|
if (registered_authenticator) {
|
|
ast_log(LOG_WARNING, "Authenticator %p is already registered. Cannot register a new one\n", registered_authenticator);
|
|
return -1;
|
|
}
|
|
registered_authenticator = auth;
|
|
ast_debug(1, "Registered SIP authenticator module %p\n", auth);
|
|
ast_module_ref(ast_module_info->self);
|
|
return 0;
|
|
}
|
|
|
|
void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth)
|
|
{
|
|
if (registered_authenticator != auth) {
|
|
ast_log(LOG_WARNING, "Trying to unregister authenticator %p but authenticator %p registered\n",
|
|
auth, registered_authenticator);
|
|
return;
|
|
}
|
|
registered_authenticator = NULL;
|
|
ast_debug(1, "Unregistered SIP authenticator %p\n", auth);
|
|
ast_module_unref(ast_module_info->self);
|
|
}
|
|
|
|
int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata)
|
|
{
|
|
if (!registered_authenticator) {
|
|
ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is not required\n");
|
|
return 0;
|
|
}
|
|
|
|
return registered_authenticator->requires_authentication(endpoint, rdata);
|
|
}
|
|
|
|
enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
|
|
pjsip_rx_data *rdata, pjsip_tx_data *tdata)
|
|
{
|
|
if (!registered_authenticator) {
|
|
ast_log(LOG_WARNING, "No SIP authenticator registered. Assuming authentication is successful\n");
|
|
return 0;
|
|
}
|
|
return registered_authenticator->check_authentication(endpoint, rdata, tdata);
|
|
}
|
|
|
|
static struct ast_sip_outbound_authenticator *registered_outbound_authenticator;
|
|
|
|
int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
|
|
{
|
|
if (registered_outbound_authenticator) {
|
|
ast_log(LOG_WARNING, "Outbound authenticator %p is already registered. Cannot register a new one\n", registered_outbound_authenticator);
|
|
return -1;
|
|
}
|
|
registered_outbound_authenticator = auth;
|
|
ast_debug(1, "Registered SIP outbound authenticator module %p\n", auth);
|
|
ast_module_ref(ast_module_info->self);
|
|
return 0;
|
|
}
|
|
|
|
void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth)
|
|
{
|
|
if (registered_outbound_authenticator != auth) {
|
|
ast_log(LOG_WARNING, "Trying to unregister outbound authenticator %p but outbound authenticator %p registered\n",
|
|
auth, registered_outbound_authenticator);
|
|
return;
|
|
}
|
|
registered_outbound_authenticator = NULL;
|
|
ast_debug(1, "Unregistered SIP outbound authenticator %p\n", auth);
|
|
ast_module_unref(ast_module_info->self);
|
|
}
|
|
|
|
int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
|
|
pjsip_transaction *tsx, pjsip_tx_data **new_request)
|
|
{
|
|
if (!registered_outbound_authenticator) {
|
|
ast_log(LOG_WARNING, "No SIP outbound authenticator registered. Cannot respond to authentication challenge\n");
|
|
return -1;
|
|
}
|
|
return registered_outbound_authenticator->create_request_with_auth(auths, challenge, tsx, new_request);
|
|
}
|
|
|
|
struct endpoint_identifier_list {
|
|
const char *name;
|
|
unsigned int priority;
|
|
struct ast_sip_endpoint_identifier *identifier;
|
|
AST_RWLIST_ENTRY(endpoint_identifier_list) list;
|
|
};
|
|
|
|
static AST_RWLIST_HEAD_STATIC(endpoint_identifiers, endpoint_identifier_list);
|
|
|
|
int ast_sip_register_endpoint_identifier_with_name(struct ast_sip_endpoint_identifier *identifier,
|
|
const char *name)
|
|
{
|
|
char *prev, *current, *identifier_order;
|
|
struct endpoint_identifier_list *iter, *id_list_item;
|
|
SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
|
|
id_list_item = ast_calloc(1, sizeof(*id_list_item));
|
|
if (!id_list_item) {
|
|
ast_log(LOG_ERROR, "Unabled to add endpoint identifier. Out of memory.\n");
|
|
return -1;
|
|
}
|
|
id_list_item->identifier = identifier;
|
|
id_list_item->name = name;
|
|
|
|
ast_debug(1, "Register endpoint identifier %s (%p)\n", name, identifier);
|
|
|
|
if (ast_strlen_zero(name)) {
|
|
/* if an identifier has no name then place in front */
|
|
AST_RWLIST_INSERT_HEAD(&endpoint_identifiers, id_list_item, list);
|
|
ast_module_ref(ast_module_info->self);
|
|
return 0;
|
|
}
|
|
|
|
/* see if the name of the identifier is in the global endpoint_identifier_order list */
|
|
identifier_order = prev = current = ast_sip_get_endpoint_identifier_order();
|
|
|
|
if (ast_strlen_zero(identifier_order)) {
|
|
id_list_item->priority = UINT_MAX;
|
|
AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_free(identifier_order);
|
|
return 0;
|
|
}
|
|
|
|
id_list_item->priority = 0;
|
|
while ((current = strchr(current, ','))) {
|
|
++id_list_item->priority;
|
|
if (!strncmp(prev, name, current - prev)) {
|
|
break;
|
|
}
|
|
prev = ++current;
|
|
}
|
|
|
|
if (!current) {
|
|
/* check to see if it is the only or last item */
|
|
if (!strcmp(prev, name)) {
|
|
++id_list_item->priority;
|
|
} else {
|
|
id_list_item->priority = UINT_MAX;
|
|
}
|
|
}
|
|
|
|
if (id_list_item->priority == UINT_MAX || AST_RWLIST_EMPTY(&endpoint_identifiers)) {
|
|
/* if not in the endpoint_identifier_order list then consider it less in
|
|
priority and add it to the end */
|
|
AST_RWLIST_INSERT_TAIL(&endpoint_identifiers, id_list_item, list);
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_free(identifier_order);
|
|
return 0;
|
|
}
|
|
|
|
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
|
|
if (id_list_item->priority < iter->priority) {
|
|
AST_RWLIST_INSERT_BEFORE_CURRENT(id_list_item, list);
|
|
break;
|
|
}
|
|
|
|
if (!AST_RWLIST_NEXT(iter, list)) {
|
|
AST_RWLIST_INSERT_AFTER(&endpoint_identifiers, iter, id_list_item, list);
|
|
break;
|
|
}
|
|
}
|
|
AST_RWLIST_TRAVERSE_SAFE_END;
|
|
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_free(identifier_order);
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
|
|
{
|
|
return ast_sip_register_endpoint_identifier_with_name(identifier, NULL);
|
|
}
|
|
|
|
void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier)
|
|
{
|
|
struct endpoint_identifier_list *iter;
|
|
SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_identifiers, iter, list) {
|
|
if (iter->identifier == identifier) {
|
|
AST_RWLIST_REMOVE_CURRENT(list);
|
|
ast_free(iter);
|
|
ast_debug(1, "Unregistered endpoint identifier %p\n", identifier);
|
|
ast_module_unref(ast_module_info->self);
|
|
break;
|
|
}
|
|
}
|
|
AST_RWLIST_TRAVERSE_SAFE_END;
|
|
}
|
|
|
|
struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata)
|
|
{
|
|
struct endpoint_identifier_list *iter;
|
|
struct ast_sip_endpoint *endpoint = NULL;
|
|
SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
|
|
ast_assert(iter->identifier->identify_endpoint != NULL);
|
|
endpoint = iter->identifier->identify_endpoint(rdata);
|
|
if (endpoint) {
|
|
break;
|
|
}
|
|
}
|
|
return endpoint;
|
|
}
|
|
|
|
static char *cli_show_endpoint_identifiers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
#define ENDPOINT_IDENTIFIER_FORMAT "%-20.20s\n"
|
|
struct endpoint_identifier_list *iter;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "pjsip show identifiers";
|
|
e->usage = "Usage: pjsip show identifiers\n"
|
|
" List all registered endpoint identifiers\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc != 3) {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
ast_cli(a->fd, ENDPOINT_IDENTIFIER_FORMAT, "Identifier Names:");
|
|
{
|
|
SCOPED_LOCK(lock, &endpoint_identifiers, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_TRAVERSE(&endpoint_identifiers, iter, list) {
|
|
ast_cli(a->fd, ENDPOINT_IDENTIFIER_FORMAT,
|
|
iter->name ? iter->name : "name not specified");
|
|
}
|
|
}
|
|
return CLI_SUCCESS;
|
|
#undef ENDPOINT_IDENTIFIER_FORMAT
|
|
}
|
|
|
|
static char *cli_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
struct ast_sip_cli_context context;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "pjsip show settings";
|
|
e->usage = "Usage: pjsip show settings\n"
|
|
" Show global and system configuration options\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
context.output_buffer = ast_str_create(256);
|
|
if (!context.output_buffer) {
|
|
ast_cli(a->fd, "Could not allocate output buffer.\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
|
|
if (sip_cli_print_global(&context) || sip_cli_print_system(&context)) {
|
|
ast_free(context.output_buffer);
|
|
ast_cli(a->fd, "Error retrieving settings.\n");
|
|
return CLI_FAILURE;
|
|
}
|
|
|
|
ast_cli(a->fd, "%s", ast_str_buffer(context.output_buffer));
|
|
ast_free(context.output_buffer);
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static struct ast_cli_entry cli_commands[] = {
|
|
AST_CLI_DEFINE(cli_show_settings, "Show global and system configuration options"),
|
|
AST_CLI_DEFINE(cli_show_endpoint_identifiers, "List registered endpoint identifiers")
|
|
};
|
|
|
|
AST_RWLIST_HEAD_STATIC(endpoint_formatters, ast_sip_endpoint_formatter);
|
|
|
|
void internal_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
|
|
{
|
|
SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_INSERT_TAIL(&endpoint_formatters, obj, next);
|
|
}
|
|
|
|
int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
|
|
{
|
|
internal_sip_register_endpoint_formatter(obj);
|
|
ast_module_ref(ast_module_info->self);
|
|
return 0;
|
|
}
|
|
|
|
int internal_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
|
|
{
|
|
struct ast_sip_endpoint_formatter *i;
|
|
SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&endpoint_formatters, i, next) {
|
|
if (i == obj) {
|
|
AST_RWLIST_REMOVE_CURRENT(next);
|
|
break;
|
|
}
|
|
}
|
|
AST_RWLIST_TRAVERSE_SAFE_END;
|
|
return i == obj ? 0 : -1;
|
|
}
|
|
|
|
void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj)
|
|
{
|
|
if (!internal_sip_unregister_endpoint_formatter(obj)) {
|
|
ast_module_unref(ast_module_info->self);
|
|
}
|
|
}
|
|
|
|
int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
|
|
struct ast_sip_ami *ami, int *count)
|
|
{
|
|
int res = 0;
|
|
struct ast_sip_endpoint_formatter *i;
|
|
SCOPED_LOCK(lock, &endpoint_formatters, AST_RWLIST_RDLOCK, AST_RWLIST_UNLOCK);
|
|
*count = 0;
|
|
AST_RWLIST_TRAVERSE(&endpoint_formatters, i, next) {
|
|
if (i->format_ami && ((res = i->format_ami(endpoint, ami)) < 0)) {
|
|
return res;
|
|
}
|
|
|
|
if (!res) {
|
|
(*count)++;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
|
|
{
|
|
return ast_pjsip_endpoint;
|
|
}
|
|
|
|
static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
|
|
{
|
|
pj_str_t tmp, local_addr;
|
|
pjsip_uri *uri;
|
|
pjsip_sip_uri *sip_uri;
|
|
pjsip_transport_type_e type = PJSIP_TRANSPORT_UNSPECIFIED;
|
|
int local_port;
|
|
char uuid_str[AST_UUID_STR_LEN];
|
|
|
|
if (ast_strlen_zero(user)) {
|
|
user = ast_uuid_generate_str(uuid_str, sizeof(uuid_str));
|
|
}
|
|
|
|
/* Parse the provided target URI so we can determine what transport it will end up using */
|
|
pj_strdup_with_null(pool, &tmp, target);
|
|
|
|
if (!(uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0)) ||
|
|
(!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
|
|
return -1;
|
|
}
|
|
|
|
sip_uri = pjsip_uri_get_uri(uri);
|
|
|
|
/* Determine the transport type to use */
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
|
|
type = PJSIP_TRANSPORT_TLS;
|
|
} else if (!sip_uri->transport_param.slen) {
|
|
type = PJSIP_TRANSPORT_UDP;
|
|
} else {
|
|
type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
|
|
}
|
|
|
|
if (type == PJSIP_TRANSPORT_UNSPECIFIED) {
|
|
return -1;
|
|
}
|
|
|
|
/* If the host is IPv6 turn the transport into an IPv6 version */
|
|
if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
|
|
type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
|
|
}
|
|
|
|
if (!ast_strlen_zero(domain)) {
|
|
from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
|
|
from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
|
|
"<sip:%s@%s%s%s>",
|
|
user,
|
|
domain,
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
|
|
return 0;
|
|
}
|
|
|
|
/* Get the local bound address for the transport that will be used when communicating with the provided URI */
|
|
if (pjsip_tpmgr_find_local_addr(pjsip_endpt_get_tpmgr(ast_sip_get_pjsip_endpoint()), pool, type, selector,
|
|
&local_addr, &local_port) != PJ_SUCCESS) {
|
|
|
|
/* If no local address can be retrieved using the transport manager use the host one */
|
|
pj_strdup(pool, &local_addr, pj_gethostname());
|
|
local_port = pjsip_transport_get_default_port_for_type(PJSIP_TRANSPORT_UDP);
|
|
}
|
|
|
|
/* If IPv6 was specified in the transport, set the proper type */
|
|
if (pj_strchr(&local_addr, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
|
|
type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
|
|
}
|
|
|
|
from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
|
|
from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
|
|
"<sip:%s@%s%.*s%s:%d%s%s>",
|
|
user,
|
|
(type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
|
|
(int)local_addr.slen,
|
|
local_addr.ptr,
|
|
(type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
|
|
local_port,
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
|
|
{
|
|
RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
|
|
const char *transport_name = endpoint->transport;
|
|
|
|
if (ast_strlen_zero(transport_name)) {
|
|
return 0;
|
|
}
|
|
|
|
transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
|
|
|
|
if (!transport || !transport->state) {
|
|
ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s' for endpoint '%s'\n",
|
|
transport_name, ast_sorcery_object_get_id(endpoint));
|
|
return -1;
|
|
}
|
|
|
|
if (transport->state->transport) {
|
|
selector->type = PJSIP_TPSELECTOR_TRANSPORT;
|
|
selector->u.transport = transport->state->transport;
|
|
} else if (transport->state->factory) {
|
|
selector->type = PJSIP_TPSELECTOR_LISTENER;
|
|
selector->u.listener = transport->state->factory;
|
|
} else if (transport->type == AST_TRANSPORT_WS || transport->type == AST_TRANSPORT_WSS) {
|
|
/* The WebSocket transport has no factory as it can not create outgoing connections, so
|
|
* even if an endpoint is locked to a WebSocket transport we let the PJSIP logic
|
|
* find the existing connection if available and use it.
|
|
*/
|
|
return 0;
|
|
} else {
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
|
|
{
|
|
pjsip_sip_uri *sip_uri;
|
|
int i = 0;
|
|
pjsip_param *param;
|
|
const pj_str_t STR_USER = { "user", 4 };
|
|
const pj_str_t STR_PHONE = { "phone", 5 };
|
|
|
|
if (!endpoint || !endpoint->usereqphone || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
|
|
return;
|
|
}
|
|
|
|
sip_uri = pjsip_uri_get_uri(uri);
|
|
|
|
if (!pj_strlen(&sip_uri->user)) {
|
|
return;
|
|
}
|
|
|
|
if (pj_strbuf(&sip_uri->user)[0] == '+') {
|
|
i = 1;
|
|
}
|
|
|
|
/* Test URI user against allowed characters in AST_DIGIT_ANY */
|
|
for (; i < pj_strlen(&sip_uri->user); i++) {
|
|
if (!strchr(AST_DIGIT_ANYNUM, pj_strbuf(&sip_uri->user)[i])) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (i < pj_strlen(&sip_uri->user)) {
|
|
return;
|
|
}
|
|
|
|
param = PJ_POOL_ALLOC_T(pool, pjsip_param);
|
|
param->name = STR_USER;
|
|
param->value = STR_PHONE;
|
|
pj_list_insert_before(&sip_uri->other_param, param);
|
|
}
|
|
|
|
pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
|
|
{
|
|
char enclosed_uri[PJSIP_MAX_URL_SIZE];
|
|
pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
|
|
pj_status_t res;
|
|
pjsip_dialog *dlg = NULL;
|
|
const char *outbound_proxy = endpoint->outbound_proxy;
|
|
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
|
|
static const pj_str_t HCONTACT = { "Contact", 7 };
|
|
|
|
snprintf(enclosed_uri, sizeof(enclosed_uri), "<%s>", uri);
|
|
pj_cstr(&remote_uri, enclosed_uri);
|
|
|
|
pj_cstr(&target_uri, uri);
|
|
|
|
res = pjsip_dlg_create_uac(pjsip_ua_instance(), &local_uri, NULL, &remote_uri, &target_uri, &dlg);
|
|
if (res != PJ_SUCCESS) {
|
|
if (res == PJSIP_EINVALIDURI) {
|
|
ast_log(LOG_ERROR, "Could not create dialog to endpoint '%s' as URI '%s' is not valid\n",
|
|
ast_sorcery_object_get_id(endpoint), uri);
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
|
|
pjsip_dlg_terminate(dlg);
|
|
return NULL;
|
|
}
|
|
|
|
if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
|
|
pjsip_dlg_terminate(dlg);
|
|
return NULL;
|
|
}
|
|
|
|
/* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
|
|
pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
|
|
dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
|
|
dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
|
|
|
|
/* If a request user has been specified and we are permitted to change it, do so */
|
|
if (!ast_strlen_zero(request_user)) {
|
|
pjsip_sip_uri *sip_uri;
|
|
|
|
if (PJSIP_URI_SCHEME_IS_SIP(dlg->target) || PJSIP_URI_SCHEME_IS_SIPS(dlg->target)) {
|
|
sip_uri = pjsip_uri_get_uri(dlg->target);
|
|
pj_strdup2(dlg->pool, &sip_uri->user, request_user);
|
|
}
|
|
if (PJSIP_URI_SCHEME_IS_SIP(dlg->remote.info->uri) || PJSIP_URI_SCHEME_IS_SIPS(dlg->remote.info->uri)) {
|
|
sip_uri = pjsip_uri_get_uri(dlg->remote.info->uri);
|
|
pj_strdup2(dlg->pool, &sip_uri->user, request_user);
|
|
}
|
|
}
|
|
|
|
/* Add the user=phone parameter if applicable */
|
|
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
|
|
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri);
|
|
|
|
/* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
|
|
dlg->sess_count++;
|
|
|
|
pjsip_dlg_set_transport(dlg, &selector);
|
|
|
|
if (!ast_strlen_zero(outbound_proxy)) {
|
|
pjsip_route_hdr route_set, *route;
|
|
static const pj_str_t ROUTE_HNAME = { "Route", 5 };
|
|
pj_str_t tmp;
|
|
|
|
pj_list_init(&route_set);
|
|
|
|
pj_strdup2_with_null(dlg->pool, &tmp, outbound_proxy);
|
|
if (!(route = pjsip_parse_hdr(dlg->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
|
|
dlg->sess_count--;
|
|
pjsip_dlg_terminate(dlg);
|
|
return NULL;
|
|
}
|
|
pj_list_insert_nodes_before(&route_set, route);
|
|
|
|
pjsip_dlg_set_route_set(dlg, &route_set);
|
|
}
|
|
|
|
dlg->sess_count--;
|
|
|
|
return dlg;
|
|
}
|
|
|
|
/*!
|
|
* \brief Determine if a SIPS Contact header is required.
|
|
*
|
|
* This uses the guideline provided in RFC 3261 Section 12.1.1 to
|
|
* determine if the Contact header must be a sips: URI.
|
|
*
|
|
* \param rdata The incoming dialog-starting request
|
|
* \retval 0 SIPS not required
|
|
* \retval 1 SIPS required
|
|
*/
|
|
static int uas_use_sips_contact(pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_rr_hdr *record_route;
|
|
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(rdata->msg_info.msg->line.req.uri)) {
|
|
return 1;
|
|
}
|
|
|
|
record_route = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_RECORD_ROUTE, NULL);
|
|
if (record_route) {
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(&record_route->name_addr)) {
|
|
return 1;
|
|
}
|
|
} else {
|
|
pjsip_contact_hdr *contact;
|
|
|
|
contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
|
|
ast_assert(contact != NULL);
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(contact->uri)) {
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status)
|
|
{
|
|
pjsip_dialog *dlg;
|
|
pj_str_t contact;
|
|
pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
|
|
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
|
|
pjsip_transport *transport;
|
|
|
|
ast_assert(status != NULL);
|
|
|
|
if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
|
|
return NULL;
|
|
}
|
|
|
|
transport = rdata->tp_info.transport;
|
|
if (selector.type == PJSIP_TPSELECTOR_TRANSPORT) {
|
|
transport = selector.u.transport;
|
|
}
|
|
type = transport->key.type;
|
|
|
|
contact.ptr = pj_pool_alloc(rdata->tp_info.pool, PJSIP_MAX_URL_SIZE);
|
|
contact.slen = pj_ansi_snprintf(contact.ptr, PJSIP_MAX_URL_SIZE,
|
|
"<%s:%s%.*s%s:%d%s%s>",
|
|
uas_use_sips_contact(rdata) ? "sips" : "sip",
|
|
(type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
|
|
(int)transport->local_name.host.slen,
|
|
transport->local_name.host.ptr,
|
|
(type & PJSIP_TRANSPORT_IPV6) ? "]" : "",
|
|
transport->local_name.port,
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
|
|
(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
|
|
|
|
*status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
|
|
if (*status != PJ_SUCCESS) {
|
|
char err[PJ_ERR_MSG_SIZE];
|
|
|
|
pj_strerror(*status, err, sizeof(err));
|
|
ast_log(LOG_ERROR, "Could not create dialog with endpoint %s. %s\n",
|
|
ast_sorcery_object_get_id(endpoint), err);
|
|
return NULL;
|
|
}
|
|
|
|
dlg->sess_count++;
|
|
pjsip_dlg_set_transport(dlg, &selector);
|
|
dlg->sess_count--;
|
|
|
|
return dlg;
|
|
}
|
|
|
|
int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
|
|
char *transport_type, const char *local_name, int local_port)
|
|
{
|
|
pj_str_t tmp;
|
|
|
|
rdata->tp_info.transport = PJ_POOL_ZALLOC_T(rdata->tp_info.pool, pjsip_transport);
|
|
if (!rdata->tp_info.transport) {
|
|
return -1;
|
|
}
|
|
|
|
ast_copy_string(rdata->pkt_info.packet, packet, sizeof(rdata->pkt_info.packet));
|
|
ast_copy_string(rdata->pkt_info.src_name, src_name, sizeof(rdata->pkt_info.src_name));
|
|
rdata->pkt_info.src_port = src_port;
|
|
|
|
pjsip_parse_rdata(packet, strlen(packet), rdata);
|
|
if (!rdata->msg_info.msg) {
|
|
return -1;
|
|
}
|
|
|
|
pj_strdup2(rdata->tp_info.pool, &rdata->msg_info.via->recvd_param, rdata->pkt_info.src_name);
|
|
rdata->msg_info.via->rport_param = -1;
|
|
|
|
rdata->tp_info.transport->key.type = pjsip_transport_get_type_from_name(pj_cstr(&tmp, transport_type));
|
|
rdata->tp_info.transport->type_name = transport_type;
|
|
pj_strdup2(rdata->tp_info.pool, &rdata->tp_info.transport->local_name.host, local_name);
|
|
rdata->tp_info.transport->local_name.port = local_port;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
|
|
static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
|
|
static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
|
|
|
|
static struct {
|
|
const char *method;
|
|
const pjsip_method *pmethod;
|
|
} methods [] = {
|
|
{ "INVITE", &pjsip_invite_method },
|
|
{ "CANCEL", &pjsip_cancel_method },
|
|
{ "ACK", &pjsip_ack_method },
|
|
{ "BYE", &pjsip_bye_method },
|
|
{ "REGISTER", &pjsip_register_method },
|
|
{ "OPTIONS", &pjsip_options_method },
|
|
{ "SUBSCRIBE", &pjsip_subscribe_method },
|
|
{ "NOTIFY", &pjsip_notify_method },
|
|
{ "PUBLISH", &pjsip_publish_method },
|
|
{ "INFO", &info_method },
|
|
{ "MESSAGE", &message_method },
|
|
};
|
|
|
|
static const pjsip_method *get_pjsip_method(const char *method)
|
|
{
|
|
int i;
|
|
for (i = 0; i < ARRAY_LEN(methods); ++i) {
|
|
if (!strcmp(method, methods[i].method)) {
|
|
return methods[i].pmethod;
|
|
}
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static int create_in_dialog_request(const pjsip_method *method, struct pjsip_dialog *dlg, pjsip_tx_data **tdata)
|
|
{
|
|
if (pjsip_dlg_create_request(dlg, method, -1, tdata) != PJ_SUCCESS) {
|
|
ast_log(LOG_WARNING, "Unable to create in-dialog request.\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata);
|
|
static pjsip_module supplement_module = {
|
|
.name = { "Out of dialog supplement hook", 29 },
|
|
.id = -1,
|
|
.priority = PJSIP_MOD_PRIORITY_APPLICATION - 1,
|
|
.on_rx_request = supplement_on_rx_request,
|
|
};
|
|
|
|
static int create_out_of_dialog_request(const pjsip_method *method, struct ast_sip_endpoint *endpoint,
|
|
const char *uri, struct ast_sip_contact *provided_contact, pjsip_tx_data **tdata)
|
|
{
|
|
RAII_VAR(struct ast_sip_contact *, contact, ao2_bump(provided_contact), ao2_cleanup);
|
|
pj_str_t remote_uri;
|
|
pj_str_t from;
|
|
pj_pool_t *pool;
|
|
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
|
|
|
|
if (ast_strlen_zero(uri)) {
|
|
if (!endpoint && (!contact || ast_strlen_zero(contact->uri))) {
|
|
ast_log(LOG_ERROR, "An endpoint and/or uri must be specified\n");
|
|
return -1;
|
|
}
|
|
|
|
if (!contact) {
|
|
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
|
|
}
|
|
if (!contact || ast_strlen_zero(contact->uri)) {
|
|
ast_log(LOG_ERROR, "Unable to retrieve contact for endpoint %s\n",
|
|
ast_sorcery_object_get_id(endpoint));
|
|
return -1;
|
|
}
|
|
|
|
pj_cstr(&remote_uri, contact->uri);
|
|
} else {
|
|
pj_cstr(&remote_uri, uri);
|
|
}
|
|
|
|
if (endpoint) {
|
|
if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
|
|
ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
|
|
ast_sorcery_object_get_id(endpoint));
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
|
|
|
|
if (!pool) {
|
|
ast_log(LOG_ERROR, "Unable to create PJLIB memory pool\n");
|
|
return -1;
|
|
}
|
|
|
|
if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
|
|
endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
|
|
ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
|
|
(int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
|
|
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
|
|
return -1;
|
|
}
|
|
|
|
if (pjsip_endpt_create_request(ast_sip_get_pjsip_endpoint(), method, &remote_uri,
|
|
&from, &remote_uri, &from, NULL, -1, NULL, tdata) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s\n",
|
|
(int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
|
|
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
|
|
return -1;
|
|
}
|
|
|
|
/* Add the user=phone parameter if applicable */
|
|
ast_sip_add_usereqphone(endpoint, (*tdata)->pool, (*tdata)->msg->line.req.uri);
|
|
|
|
/* If an outbound proxy is specified on the endpoint apply it to this request */
|
|
if (endpoint && !ast_strlen_zero(endpoint->outbound_proxy) &&
|
|
ast_sip_set_outbound_proxy((*tdata), endpoint->outbound_proxy)) {
|
|
ast_log(LOG_ERROR, "Unable to apply outbound proxy on request %.*s to endpoint %s\n",
|
|
(int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
|
|
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
|
|
return -1;
|
|
}
|
|
|
|
ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
|
|
|
|
/* We can release this pool since request creation copied all the necessary
|
|
* data into the outbound request's pool
|
|
*/
|
|
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
|
|
struct ast_sip_endpoint *endpoint, const char *uri,
|
|
struct ast_sip_contact *contact, pjsip_tx_data **tdata)
|
|
{
|
|
const pjsip_method *pmethod = get_pjsip_method(method);
|
|
|
|
if (!pmethod) {
|
|
ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
|
|
return -1;
|
|
}
|
|
|
|
if (dlg) {
|
|
return create_in_dialog_request(pmethod, dlg, tdata);
|
|
} else {
|
|
return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
|
|
}
|
|
}
|
|
|
|
AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
|
|
|
|
int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
|
|
{
|
|
struct ast_sip_supplement *iter;
|
|
int inserted = 0;
|
|
SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
|
|
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
|
|
if (iter->priority > supplement->priority) {
|
|
AST_RWLIST_INSERT_BEFORE_CURRENT(supplement, next);
|
|
inserted = 1;
|
|
break;
|
|
}
|
|
}
|
|
AST_RWLIST_TRAVERSE_SAFE_END;
|
|
|
|
if (!inserted) {
|
|
AST_RWLIST_INSERT_TAIL(&supplements, supplement, next);
|
|
}
|
|
ast_module_ref(ast_module_info->self);
|
|
return 0;
|
|
}
|
|
|
|
void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
|
|
{
|
|
struct ast_sip_supplement *iter;
|
|
SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
|
|
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
|
|
if (supplement == iter) {
|
|
AST_RWLIST_REMOVE_CURRENT(next);
|
|
ast_module_unref(ast_module_info->self);
|
|
break;
|
|
}
|
|
}
|
|
AST_RWLIST_TRAVERSE_SAFE_END;
|
|
}
|
|
|
|
static int send_in_dialog_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg)
|
|
{
|
|
if (pjsip_dlg_send_request(dlg, tdata, -1, NULL) != PJ_SUCCESS) {
|
|
ast_log(LOG_WARNING, "Unable to send in-dialog request.\n");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
|
|
{
|
|
pj_str_t method;
|
|
|
|
if (ast_strlen_zero(supplement_method)) {
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
pj_cstr(&method, supplement_method);
|
|
|
|
return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
|
|
}
|
|
|
|
/*! Maximum number of challenges before assuming that we are in a loop */
|
|
#define MAX_RX_CHALLENGES 10
|
|
|
|
/*! \brief Structure to hold information about an outbound request */
|
|
struct send_request_data {
|
|
/*! The endpoint associated with this request */
|
|
struct ast_sip_endpoint *endpoint;
|
|
/*! Information to be provided to the callback upon receipt of a response */
|
|
void *token;
|
|
/*! The callback to be called upon receipt of a response */
|
|
void (*callback)(void *token, pjsip_event *e);
|
|
/*! Number of challenges received. */
|
|
unsigned int challenge_count;
|
|
};
|
|
|
|
static void send_request_data_destroy(void *obj)
|
|
{
|
|
struct send_request_data *req_data = obj;
|
|
|
|
ao2_cleanup(req_data->endpoint);
|
|
}
|
|
|
|
static struct send_request_data *send_request_data_alloc(struct ast_sip_endpoint *endpoint,
|
|
void *token, void (*callback)(void *token, pjsip_event *e))
|
|
{
|
|
struct send_request_data *req_data;
|
|
|
|
req_data = ao2_alloc_options(sizeof(*req_data), send_request_data_destroy,
|
|
AO2_ALLOC_OPT_LOCK_NOLOCK);
|
|
if (!req_data) {
|
|
return NULL;
|
|
}
|
|
|
|
req_data->endpoint = ao2_bump(endpoint);
|
|
req_data->token = token;
|
|
req_data->callback = callback;
|
|
|
|
return req_data;
|
|
}
|
|
|
|
struct send_request_wrapper {
|
|
/*! Information to be provided to the callback upon receipt of a response */
|
|
void *token;
|
|
/*! The callback to be called upon receipt of a response */
|
|
void (*callback)(void *token, pjsip_event *e);
|
|
/*! Non-zero when the callback is called. */
|
|
unsigned int cb_called;
|
|
};
|
|
|
|
static void endpt_send_request_wrapper(void *token, pjsip_event *e)
|
|
{
|
|
struct send_request_wrapper *req_wrapper = token;
|
|
|
|
req_wrapper->cb_called = 1;
|
|
if (req_wrapper->callback) {
|
|
req_wrapper->callback(req_wrapper->token, e);
|
|
}
|
|
ao2_ref(req_wrapper, -1);
|
|
}
|
|
|
|
static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
|
|
pjsip_tx_data *tdata, pj_int32_t timeout, void *token, pjsip_endpt_send_callback cb)
|
|
{
|
|
struct send_request_wrapper *req_wrapper;
|
|
pj_status_t ret_val;
|
|
|
|
/* Create wrapper to detect if the callback was actually called on an error. */
|
|
req_wrapper = ao2_alloc_options(sizeof(*req_wrapper), NULL,
|
|
AO2_ALLOC_OPT_LOCK_NOLOCK);
|
|
if (!req_wrapper) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
return PJ_ENOMEM;
|
|
}
|
|
req_wrapper->token = token;
|
|
req_wrapper->callback = cb;
|
|
|
|
ao2_ref(req_wrapper, +1);
|
|
ret_val = pjsip_endpt_send_request(ast_sip_get_pjsip_endpoint(), tdata, timeout,
|
|
req_wrapper, endpt_send_request_wrapper);
|
|
if (ret_val != PJ_SUCCESS) {
|
|
char errmsg[PJ_ERR_MSG_SIZE];
|
|
|
|
/* Complain of failure to send the request. */
|
|
pj_strerror(ret_val, errmsg, sizeof(errmsg));
|
|
ast_log(LOG_ERROR, "Error %d '%s' sending %.*s request to endpoint %s\n",
|
|
(int) ret_val, errmsg, (int) pj_strlen(&tdata->msg->line.req.method.name),
|
|
pj_strbuf(&tdata->msg->line.req.method.name),
|
|
endpoint ? ast_sorcery_object_get_id(endpoint) : "<unknown>");
|
|
|
|
/* Was the callback called? */
|
|
if (req_wrapper->cb_called) {
|
|
/*
|
|
* Yes so we cannot report any error. The callback
|
|
* has already freed any resources associated with
|
|
* token.
|
|
*/
|
|
ret_val = PJ_SUCCESS;
|
|
} else {
|
|
/* No and it is not expected to ever be called. */
|
|
ao2_ref(req_wrapper, -1);
|
|
}
|
|
}
|
|
ao2_ref(req_wrapper, -1);
|
|
return ret_val;
|
|
}
|
|
|
|
static void send_request_cb(void *token, pjsip_event *e)
|
|
{
|
|
struct send_request_data *req_data = token;
|
|
pjsip_transaction *tsx;
|
|
pjsip_rx_data *challenge;
|
|
pjsip_tx_data *tdata;
|
|
struct ast_sip_supplement *supplement;
|
|
struct ast_sip_endpoint *endpoint;
|
|
int res;
|
|
|
|
switch(e->body.tsx_state.type) {
|
|
case PJSIP_EVENT_TRANSPORT_ERROR:
|
|
case PJSIP_EVENT_TIMER:
|
|
break;
|
|
case PJSIP_EVENT_RX_MSG:
|
|
challenge = e->body.tsx_state.src.rdata;
|
|
|
|
/*
|
|
* Call any supplements that want to know about a response
|
|
* with any received data.
|
|
*/
|
|
AST_RWLIST_RDLOCK(&supplements);
|
|
AST_LIST_TRAVERSE(&supplements, supplement, next) {
|
|
if (supplement->incoming_response
|
|
&& does_method_match(&challenge->msg_info.cseq->method.name,
|
|
supplement->method)) {
|
|
supplement->incoming_response(req_data->endpoint, challenge);
|
|
}
|
|
}
|
|
AST_RWLIST_UNLOCK(&supplements);
|
|
|
|
/* Resend the request with a challenge response if we are challenged. */
|
|
tsx = e->body.tsx_state.tsx;
|
|
endpoint = ao2_bump(req_data->endpoint);
|
|
res = (tsx->status_code == 401 || tsx->status_code == 407)
|
|
&& endpoint
|
|
&& ++req_data->challenge_count < MAX_RX_CHALLENGES /* Not in a challenge loop */
|
|
&& !ast_sip_create_request_with_auth(&endpoint->outbound_auths,
|
|
challenge, tsx, &tdata)
|
|
&& endpt_send_request(endpoint, tdata, -1, req_data, send_request_cb)
|
|
== PJ_SUCCESS;
|
|
ao2_cleanup(endpoint);
|
|
if (res) {
|
|
/*
|
|
* Request with challenge response sent.
|
|
* Passed our req_data ref to the new request.
|
|
*/
|
|
return;
|
|
}
|
|
break;
|
|
default:
|
|
ast_log(LOG_ERROR, "Unexpected PJSIP event %u\n", e->body.tsx_state.type);
|
|
break;
|
|
}
|
|
|
|
if (req_data->callback) {
|
|
req_data->callback(req_data->token, e);
|
|
}
|
|
ao2_ref(req_data, -1);
|
|
}
|
|
|
|
static int send_out_of_dialog_request(pjsip_tx_data *tdata, struct ast_sip_endpoint *endpoint,
|
|
void *token, void (*callback)(void *token, pjsip_event *e))
|
|
{
|
|
struct ast_sip_supplement *supplement;
|
|
struct send_request_data *req_data;
|
|
struct ast_sip_contact *contact;
|
|
|
|
req_data = send_request_data_alloc(endpoint, token, callback);
|
|
if (!req_data) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
return -1;
|
|
}
|
|
|
|
contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
|
|
|
|
AST_RWLIST_RDLOCK(&supplements);
|
|
AST_LIST_TRAVERSE(&supplements, supplement, next) {
|
|
if (supplement->outgoing_request
|
|
&& does_method_match(&tdata->msg->line.req.method.name, supplement->method)) {
|
|
supplement->outgoing_request(endpoint, contact, tdata);
|
|
}
|
|
}
|
|
AST_RWLIST_UNLOCK(&supplements);
|
|
|
|
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
|
|
ao2_cleanup(contact);
|
|
|
|
if (endpt_send_request(endpoint, tdata, -1, req_data, send_request_cb)
|
|
!= PJ_SUCCESS) {
|
|
ao2_cleanup(req_data);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
|
|
struct ast_sip_endpoint *endpoint, void *token,
|
|
void (*callback)(void *token, pjsip_event *e))
|
|
{
|
|
ast_assert(tdata->msg->type == PJSIP_REQUEST_MSG);
|
|
|
|
if (dlg) {
|
|
return send_in_dialog_request(tdata, dlg);
|
|
} else {
|
|
return send_out_of_dialog_request(tdata, endpoint, token, callback);
|
|
}
|
|
}
|
|
|
|
int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy)
|
|
{
|
|
pjsip_route_hdr *route;
|
|
static const pj_str_t ROUTE_HNAME = { "Route", 5 };
|
|
pj_str_t tmp;
|
|
|
|
pj_strdup2_with_null(tdata->pool, &tmp, proxy);
|
|
if (!(route = pjsip_parse_hdr(tdata->pool, &ROUTE_HNAME, tmp.ptr, tmp.slen, NULL))) {
|
|
return -1;
|
|
}
|
|
|
|
pj_list_insert_nodes_before(&tdata->msg->hdr, (pjsip_hdr*)route);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
|
|
{
|
|
pj_str_t hdr_name;
|
|
pj_str_t hdr_value;
|
|
pjsip_generic_string_hdr *hdr;
|
|
|
|
pj_cstr(&hdr_name, name);
|
|
pj_cstr(&hdr_value, value);
|
|
|
|
hdr = pjsip_generic_string_hdr_create(tdata->pool, &hdr_name, &hdr_value);
|
|
|
|
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *) hdr);
|
|
return 0;
|
|
}
|
|
|
|
static pjsip_msg_body *ast_body_to_pjsip_body(pj_pool_t *pool, const struct ast_sip_body *body)
|
|
{
|
|
pj_str_t type;
|
|
pj_str_t subtype;
|
|
pj_str_t body_text;
|
|
|
|
pj_cstr(&type, body->type);
|
|
pj_cstr(&subtype, body->subtype);
|
|
pj_cstr(&body_text, body->body_text);
|
|
|
|
return pjsip_msg_body_create(pool, &type, &subtype, &body_text);
|
|
}
|
|
|
|
int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
|
|
{
|
|
pjsip_msg_body *pjsip_body = ast_body_to_pjsip_body(tdata->pool, body);
|
|
tdata->msg->body = pjsip_body;
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies)
|
|
{
|
|
int i;
|
|
/* NULL for type and subtype automatically creates "multipart/mixed" */
|
|
pjsip_msg_body *body = pjsip_multipart_create(tdata->pool, NULL, NULL);
|
|
|
|
for (i = 0; i < num_bodies; ++i) {
|
|
pjsip_multipart_part *part = pjsip_multipart_create_part(tdata->pool);
|
|
part->body = ast_body_to_pjsip_body(tdata->pool, bodies[i]);
|
|
pjsip_multipart_add_part(tdata->pool, body, part);
|
|
}
|
|
|
|
tdata->msg->body = body;
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text)
|
|
{
|
|
size_t combined_size = strlen(body_text) + tdata->msg->body->len;
|
|
struct ast_str *body_buffer = ast_str_alloca(combined_size);
|
|
|
|
ast_str_set(&body_buffer, 0, "%.*s%s", (int) tdata->msg->body->len, (char *) tdata->msg->body->data, body_text);
|
|
|
|
tdata->msg->body->data = pj_pool_alloc(tdata->pool, combined_size);
|
|
pj_memcpy(tdata->msg->body->data, ast_str_buffer(body_buffer), combined_size);
|
|
tdata->msg->body->len = combined_size;
|
|
|
|
return 0;
|
|
}
|
|
|
|
struct ast_taskprocessor *ast_sip_create_serializer(void)
|
|
{
|
|
struct ast_taskprocessor *serializer;
|
|
char name[AST_UUID_STR_LEN];
|
|
|
|
ast_uuid_generate_str(name, sizeof(name));
|
|
|
|
serializer = ast_threadpool_serializer(name, sip_threadpool);
|
|
if (!serializer) {
|
|
return NULL;
|
|
}
|
|
return serializer;
|
|
}
|
|
|
|
int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
|
|
{
|
|
if (serializer) {
|
|
return ast_taskprocessor_push(serializer, sip_task, task_data);
|
|
} else {
|
|
return ast_threadpool_push(sip_threadpool, sip_task, task_data);
|
|
}
|
|
}
|
|
|
|
struct sync_task_data {
|
|
ast_mutex_t lock;
|
|
ast_cond_t cond;
|
|
int complete;
|
|
int fail;
|
|
int (*task)(void *);
|
|
void *task_data;
|
|
};
|
|
|
|
static int sync_task(void *data)
|
|
{
|
|
struct sync_task_data *std = data;
|
|
int ret;
|
|
|
|
std->fail = std->task(std->task_data);
|
|
|
|
/*
|
|
* Once we unlock std->lock after signaling, we cannot access
|
|
* std again. The thread waiting within
|
|
* ast_sip_push_task_synchronous() is free to continue and
|
|
* release its local variable (std).
|
|
*/
|
|
ast_mutex_lock(&std->lock);
|
|
std->complete = 1;
|
|
ast_cond_signal(&std->cond);
|
|
ret = std->fail;
|
|
ast_mutex_unlock(&std->lock);
|
|
return ret;
|
|
}
|
|
|
|
int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
|
|
{
|
|
/* This method is an onion */
|
|
struct sync_task_data std;
|
|
|
|
if (ast_sip_thread_is_servant()) {
|
|
return sip_task(task_data);
|
|
}
|
|
|
|
memset(&std, 0, sizeof(std));
|
|
ast_mutex_init(&std.lock);
|
|
ast_cond_init(&std.cond, NULL);
|
|
std.task = sip_task;
|
|
std.task_data = task_data;
|
|
|
|
if (serializer) {
|
|
if (ast_taskprocessor_push(serializer, sync_task, &std)) {
|
|
ast_mutex_destroy(&std.lock);
|
|
ast_cond_destroy(&std.cond);
|
|
return -1;
|
|
}
|
|
} else {
|
|
if (ast_threadpool_push(sip_threadpool, sync_task, &std)) {
|
|
ast_mutex_destroy(&std.lock);
|
|
ast_cond_destroy(&std.cond);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
ast_mutex_lock(&std.lock);
|
|
while (!std.complete) {
|
|
ast_cond_wait(&std.cond, &std.lock);
|
|
}
|
|
ast_mutex_unlock(&std.lock);
|
|
|
|
ast_mutex_destroy(&std.lock);
|
|
ast_cond_destroy(&std.cond);
|
|
return std.fail;
|
|
}
|
|
|
|
void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
|
|
{
|
|
size_t chars_to_copy = MIN(size - 1, pj_strlen(src));
|
|
memcpy(dest, pj_strbuf(src), chars_to_copy);
|
|
dest[chars_to_copy] = '\0';
|
|
}
|
|
|
|
int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype)
|
|
{
|
|
pjsip_media_type compare;
|
|
|
|
if (!content_type) {
|
|
return 0;
|
|
}
|
|
|
|
pjsip_media_type_init2(&compare, type, subtype);
|
|
|
|
return pjsip_media_type_cmp(content_type, &compare, 0) ? 0 : -1;
|
|
}
|
|
|
|
pj_caching_pool caching_pool;
|
|
pj_pool_t *memory_pool;
|
|
pj_thread_t *monitor_thread;
|
|
static int monitor_continue;
|
|
|
|
static void *monitor_thread_exec(void *endpt)
|
|
{
|
|
while (monitor_continue) {
|
|
const pj_time_val delay = {0, 10};
|
|
pjsip_endpt_handle_events(ast_pjsip_endpoint, &delay);
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static void stop_monitor_thread(void)
|
|
{
|
|
monitor_continue = 0;
|
|
pj_thread_join(monitor_thread);
|
|
}
|
|
|
|
AST_THREADSTORAGE(pj_thread_storage);
|
|
AST_THREADSTORAGE(servant_id_storage);
|
|
#define SIP_SERVANT_ID 0x5E2F1D
|
|
|
|
static void sip_thread_start(void)
|
|
{
|
|
pj_thread_desc *desc;
|
|
pj_thread_t *thread;
|
|
uint32_t *servant_id;
|
|
|
|
servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
|
|
if (!servant_id) {
|
|
ast_log(LOG_ERROR, "Could not set SIP servant ID in thread-local storage.\n");
|
|
return;
|
|
}
|
|
*servant_id = SIP_SERVANT_ID;
|
|
|
|
desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
|
|
if (!desc) {
|
|
ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
|
|
return;
|
|
}
|
|
pj_bzero(*desc, sizeof(*desc));
|
|
|
|
if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Couldn't register thread with PJLIB.\n");
|
|
}
|
|
}
|
|
|
|
int ast_sip_thread_is_servant(void)
|
|
{
|
|
uint32_t *servant_id;
|
|
|
|
if (monitor_thread &&
|
|
pthread_self() == *(pthread_t *)pj_thread_get_os_handle(monitor_thread)) {
|
|
return 1;
|
|
}
|
|
|
|
servant_id = ast_threadstorage_get(&servant_id_storage, sizeof(*servant_id));
|
|
if (!servant_id) {
|
|
return 0;
|
|
}
|
|
|
|
return *servant_id == SIP_SERVANT_ID;
|
|
}
|
|
|
|
void *ast_sip_dict_get(void *ht, const char *key)
|
|
{
|
|
unsigned int hval = 0;
|
|
|
|
if (!ht) {
|
|
return NULL;
|
|
}
|
|
|
|
return pj_hash_get(ht, key, PJ_HASH_KEY_STRING, &hval);
|
|
}
|
|
|
|
void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
|
|
const char *key, void *val)
|
|
{
|
|
if (!ht) {
|
|
ht = pj_hash_create(pool, 11);
|
|
}
|
|
|
|
pj_hash_set(pool, ht, key, PJ_HASH_KEY_STRING, 0, val);
|
|
|
|
return ht;
|
|
}
|
|
|
|
static pj_bool_t supplement_on_rx_request(pjsip_rx_data *rdata)
|
|
{
|
|
struct ast_sip_supplement *supplement;
|
|
|
|
if (pjsip_rdata_get_dlg(rdata)) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
AST_RWLIST_RDLOCK(&supplements);
|
|
AST_LIST_TRAVERSE(&supplements, supplement, next) {
|
|
if (supplement->incoming_request
|
|
&& does_method_match(&rdata->msg_info.msg->line.req.method.name, supplement->method)) {
|
|
struct ast_sip_endpoint *endpoint;
|
|
|
|
endpoint = ast_pjsip_rdata_get_endpoint(rdata);
|
|
supplement->incoming_request(endpoint, rdata);
|
|
ao2_cleanup(endpoint);
|
|
}
|
|
}
|
|
AST_RWLIST_UNLOCK(&supplements);
|
|
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
static void supplement_outgoing_response(pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
|
|
{
|
|
struct ast_sip_supplement *supplement;
|
|
pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
|
|
struct ast_sip_contact *contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
|
|
|
|
AST_RWLIST_RDLOCK(&supplements);
|
|
AST_LIST_TRAVERSE(&supplements, supplement, next) {
|
|
if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
|
|
supplement->outgoing_response(sip_endpoint, contact, tdata);
|
|
}
|
|
}
|
|
AST_RWLIST_UNLOCK(&supplements);
|
|
|
|
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
|
|
ao2_cleanup(contact);
|
|
}
|
|
|
|
int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
|
|
{
|
|
supplement_outgoing_response(tdata, sip_endpoint);
|
|
|
|
return pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
|
|
}
|
|
|
|
int ast_sip_send_stateful_response(pjsip_rx_data *rdata, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint)
|
|
{
|
|
pjsip_transaction *tsx;
|
|
|
|
if (pjsip_tsx_create_uas(NULL, rdata, &tsx) != PJ_SUCCESS) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
return -1;
|
|
}
|
|
pjsip_tsx_recv_msg(tsx, rdata);
|
|
|
|
supplement_outgoing_response(tdata, sip_endpoint);
|
|
|
|
if (pjsip_tsx_send_msg(tsx, tdata) != PJ_SUCCESS) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
|
|
struct ast_sip_contact *contact, pjsip_tx_data **tdata)
|
|
{
|
|
int res = pjsip_endpt_create_response(ast_sip_get_pjsip_endpoint(), rdata, st_code, NULL, tdata);
|
|
|
|
if (!res) {
|
|
ast_sip_mod_data_set((*tdata)->pool, (*tdata)->mod_data, supplement_module.id, MOD_DATA_CONTACT, ao2_bump(contact));
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void remove_request_headers(pjsip_endpoint *endpt)
|
|
{
|
|
const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
|
|
pjsip_hdr *iter = request_headers->next;
|
|
|
|
while (iter != request_headers) {
|
|
pjsip_hdr *to_erase = iter;
|
|
iter = iter->next;
|
|
pj_list_erase(to_erase);
|
|
}
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Reload configuration within a PJSIP thread
|
|
*/
|
|
static int reload_configuration_task(void *obj)
|
|
{
|
|
ast_res_pjsip_reload_configuration();
|
|
ast_res_pjsip_init_options_handling(1);
|
|
ast_sip_initialize_dns();
|
|
return 0;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
/* The third parameter is just copied from
|
|
* example code from PJLIB. This can be adjusted
|
|
* if necessary.
|
|
*/
|
|
pj_status_t status;
|
|
struct ast_threadpool_options options;
|
|
|
|
if (pj_init() != PJ_SUCCESS) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (pjlib_util_init() != PJ_SUCCESS) {
|
|
pj_shutdown();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
pj_caching_pool_init(&caching_pool, NULL, 1024 * 1024);
|
|
if (pjsip_endpt_create(&caching_pool.factory, "SIP", &ast_pjsip_endpoint) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
/* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
|
|
* we need to stop PJSIP from doing it automatically
|
|
*/
|
|
remove_request_headers(ast_pjsip_endpoint);
|
|
|
|
memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
|
|
if (!memory_pool) {
|
|
ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (ast_sip_initialize_system()) {
|
|
ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
sip_get_threadpool_options(&options);
|
|
options.thread_start = sip_thread_start;
|
|
sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
|
|
if (!sip_threadpool) {
|
|
ast_log(LOG_ERROR, "Failed to create SIP threadpool. Aborting load\n");
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_sip_initialize_dns();
|
|
|
|
pjsip_tsx_layer_init_module(ast_pjsip_endpoint);
|
|
pjsip_ua_init_module(ast_pjsip_endpoint, NULL);
|
|
|
|
monitor_continue = 1;
|
|
status = pj_thread_create(memory_pool, "SIP", (pj_thread_proc *) &monitor_thread_exec,
|
|
NULL, PJ_THREAD_DEFAULT_STACK_SIZE * 2, 0, &monitor_thread);
|
|
if (status != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Failed to start SIP monitor thread. Aborting load\n");
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_sip_initialize_global_headers();
|
|
|
|
if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
|
|
ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
|
|
ast_sip_destroy_global_headers();
|
|
stop_monitor_thread();
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (ast_sip_initialize_distributor()) {
|
|
ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
|
|
ast_res_pjsip_destroy_configuration();
|
|
ast_sip_destroy_global_headers();
|
|
stop_monitor_thread();
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (internal_sip_register_service(&supplement_module)) {
|
|
ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
|
|
ast_sip_destroy_distributor();
|
|
ast_res_pjsip_destroy_configuration();
|
|
ast_sip_destroy_global_headers();
|
|
stop_monitor_thread();
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (internal_sip_initialize_outbound_authentication()) {
|
|
ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
|
|
internal_sip_unregister_service(&supplement_module);
|
|
ast_sip_destroy_distributor();
|
|
ast_res_pjsip_destroy_configuration();
|
|
ast_sip_destroy_global_headers();
|
|
stop_monitor_thread();
|
|
ast_sip_destroy_system();
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
pjsip_endpt_destroy(ast_pjsip_endpoint);
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_res_pjsip_init_options_handling(0);
|
|
ast_cli_register_multiple(cli_commands, ARRAY_LEN(cli_commands));
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int reload_module(void)
|
|
{
|
|
/*
|
|
* We must wait for the reload to complete so multiple
|
|
* reloads cannot happen at the same time.
|
|
*/
|
|
if (ast_sip_push_task_synchronous(NULL, reload_configuration_task, NULL)) {
|
|
ast_log(LOG_WARNING, "Failed to reload PJSIP\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int unload_pjsip(void *data)
|
|
{
|
|
ast_cli_unregister_multiple(cli_commands, ARRAY_LEN(cli_commands));
|
|
ast_res_pjsip_cleanup_options_handling();
|
|
internal_sip_destroy_outbound_authentication();
|
|
ast_sip_destroy_distributor();
|
|
ast_res_pjsip_destroy_configuration();
|
|
ast_sip_destroy_system();
|
|
ast_sip_destroy_global_headers();
|
|
internal_sip_unregister_service(&supplement_module);
|
|
if (monitor_thread) {
|
|
stop_monitor_thread();
|
|
}
|
|
if (memory_pool) {
|
|
pj_pool_release(memory_pool);
|
|
memory_pool = NULL;
|
|
}
|
|
ast_pjsip_endpoint = NULL;
|
|
pj_caching_pool_destroy(&caching_pool);
|
|
pj_shutdown();
|
|
return 0;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
/* The thread this is called from cannot call PJSIP/PJLIB functions,
|
|
* so we have to push the work to the threadpool to handle
|
|
*/
|
|
ast_sip_push_task_synchronous(NULL, unload_pjsip, NULL);
|
|
|
|
ast_threadpool_shutdown(sip_threadpool);
|
|
|
|
ast_sip_destroy_cli();
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Basic SIP resource",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
|
|
);
|