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- /*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2013, Digium, Inc.
- *
- * Mark Michelson <mmichelson@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
- #include "asterisk.h"
- #include <pjsip.h>
- /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
- #include <pjsip_simple.h>
- #include <pjsip/sip_transaction.h>
- #include <pj/timer.h>
- #include <pjlib.h>
- #include <pjmedia/errno.h>
- #include "asterisk/res_pjsip.h"
- #include "res_pjsip/include/res_pjsip_private.h"
- #include "asterisk/linkedlists.h"
- #include "asterisk/logger.h"
- #include "asterisk/lock.h"
- #include "asterisk/utils.h"
- #include "asterisk/astobj2.h"
- #include "asterisk/module.h"
- #include "asterisk/threadpool.h"
- #include "asterisk/taskprocessor.h"
- #include "asterisk/uuid.h"
- #include "asterisk/sorcery.h"
- #include "asterisk/file.h"
- #include "asterisk/cli.h"
- #include "asterisk/res_pjsip_cli.h"
- #include "asterisk/test.h"
- #include "asterisk/res_pjsip_presence_xml.h"
- #include "asterisk/res_pjproject.h"
- /*** MODULEINFO
- <depend>pjproject</depend>
- <depend>res_pjproject</depend>
- <depend>res_sorcery_config</depend>
- <depend>res_sorcery_memory</depend>
- <depend>res_sorcery_astdb</depend>
- <use type="module">res_statsd</use>
- <support_level>core</support_level>
- ***/
- /*** DOCUMENTATION
- <configInfo name="res_pjsip" language="en_US">
- <synopsis>SIP Resource using PJProject</synopsis>
- <configFile name="pjsip.conf">
- <configObject name="endpoint">
- <synopsis>Endpoint</synopsis>
- <description><para>
- The <emphasis>Endpoint</emphasis> is the primary configuration object.
- It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
- dialable entries of their own. Communication with another SIP device is
- accomplished via Addresses of Record (AoRs) which have one or more
- contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to
- use a <literal>transport</literal> will default to first transport found
- in <filename>pjsip.conf</filename> that matches its type.
- </para>
- <para>Example: An Endpoint has been configured with no transport.
- When it comes time to call an AoR, PJSIP will find the
- first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
- will use the first IPv6 transport and try to send the request.
- </para>
- <para>If the anonymous endpoint identifier is in use an endpoint with the name
- "anonymous@domain" will be searched for as a last resort. If this is not found
- it will fall back to searching for "anonymous". If neither endpoints are found
- the anonymous endpoint identifier will not return an endpoint and anonymous
- calling will not be possible.
- </para>
- </description>
- <configOption name="100rel" default="yes">
- <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
- <description>
- <enumlist>
- <enum name="no" />
- <enum name="required" />
- <enum name="yes" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="aggregate_mwi" default="yes">
- <synopsis>Condense MWI notifications into a single NOTIFY.</synopsis>
- <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
- waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
- individual NOTIFYs are sent for each mailbox.</para></description>
- </configOption>
- <configOption name="allow">
- <synopsis>Media Codec(s) to allow</synopsis>
- </configOption>
- <configOption name="allow_overlap" default="yes">
- <synopsis>Enable RFC3578 overlap dialing support.</synopsis>
- </configOption>
- <configOption name="aors">
- <synopsis>AoR(s) to be used with the endpoint</synopsis>
- <description><para>
- List of comma separated AoRs that the endpoint should be associated with.
- </para></description>
- </configOption>
- <configOption name="auth">
- <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
- <description><para>
- This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
- in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
- </para><para>
- Endpoints without an authentication object
- configured will allow connections without verification.</para>
- <note><para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses. See the auth realm description for details.
- </para></note>
- </description>
- </configOption>
- <configOption name="callerid">
- <synopsis>CallerID information for the endpoint</synopsis>
- <description><para>
- Must be in the format <literal>Name <Number></literal>,
- or only <literal><Number></literal>.
- </para></description>
- </configOption>
- <configOption name="callerid_privacy">
- <synopsis>Default privacy level</synopsis>
- <description>
- <enumlist>
- <enum name="allowed_not_screened" />
- <enum name="allowed_passed_screen" />
- <enum name="allowed_failed_screen" />
- <enum name="allowed" />
- <enum name="prohib_not_screened" />
- <enum name="prohib_passed_screen" />
- <enum name="prohib_failed_screen" />
- <enum name="prohib" />
- <enum name="unavailable" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="callerid_tag">
- <synopsis>Internal id_tag for the endpoint</synopsis>
- </configOption>
- <configOption name="context">
- <synopsis>Dialplan context for inbound sessions</synopsis>
- </configOption>
- <configOption name="direct_media_glare_mitigation" default="none">
- <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
- <description>
- <para>
- This setting attempts to avoid creating INVITE glare scenarios
- by disabling direct media reINVITEs in one direction thereby allowing
- designated servers (according to this option) to initiate direct
- media reINVITEs without contention and significantly reducing call
- setup time.
- </para>
- <para>
- A more detailed description of how this option functions can be found on
- the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
- </para>
- <enumlist>
- <enum name="none" />
- <enum name="outgoing" />
- <enum name="incoming" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="direct_media_method" default="invite">
- <synopsis>Direct Media method type</synopsis>
- <description>
- <para>Method for setting up Direct Media between endpoints.</para>
- <enumlist>
- <enum name="invite" />
- <enum name="reinvite">
- <para>Alias for the <literal>invite</literal> value.</para>
- </enum>
- <enum name="update" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="trust_connected_line">
- <synopsis>Accept Connected Line updates from this endpoint</synopsis>
- </configOption>
- <configOption name="send_connected_line">
- <synopsis>Send Connected Line updates to this endpoint</synopsis>
- </configOption>
- <configOption name="connected_line_method" default="invite">
- <synopsis>Connected line method type</synopsis>
- <description>
- <para>Method used when updating connected line information.</para>
- <enumlist>
- <enum name="invite">
- <para>When set to <literal>invite</literal>, check the remote's Allow header and
- if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP
- renegotiation. If UPDATE is not Allowed, send INVITE.</para>
- </enum>
- <enum name="reinvite">
- <para>Alias for the <literal>invite</literal> value.</para>
- </enum>
- <enum name="update">
- <para>If set to <literal>update</literal>, send UPDATE regardless of what the remote
- Allows. </para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="direct_media" default="yes">
- <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
- </configOption>
- <configOption name="disable_direct_media_on_nat" default="no">
- <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
- </configOption>
- <configOption name="disallow">
- <synopsis>Media Codec(s) to disallow</synopsis>
- </configOption>
- <configOption name="dtmf_mode" default="rfc4733">
- <synopsis>DTMF mode</synopsis>
- <description>
- <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
- <enumlist>
- <enum name="rfc4733">
- <para>DTMF is sent out of band of the main audio stream. This
- supercedes the older <emphasis>RFC-2833</emphasis> used within
- the older <literal>chan_sip</literal>.</para>
- </enum>
- <enum name="inband">
- <para>DTMF is sent as part of audio stream.</para>
- </enum>
- <enum name="info">
- <para>DTMF is sent as SIP INFO packets.</para>
- </enum>
- <enum name="auto">
- <para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
- </enum>
- <enum name="auto_info">
- <para>DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.</para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="media_address">
- <synopsis>IP address used in SDP for media handling</synopsis>
- <description><para>
- At the time of SDP creation, the IP address defined here will be used as
- the media address for individual streams in the SDP.
- </para>
- <note><para>
- Be aware that the <literal>external_media_address</literal> option, set in Transport
- configuration, can also affect the final media address used in the SDP.
- </para></note>
- </description>
- </configOption>
- <configOption name="bind_rtp_to_media_address">
- <synopsis>Bind the RTP instance to the media_address</synopsis>
- <description><para>
- If media_address is specified, this option causes the RTP instance to be bound to the
- specified ip address which causes the packets to be sent from that address.
- </para>
- </description>
- </configOption>
- <configOption name="force_rport" default="yes">
- <synopsis>Force use of return port</synopsis>
- </configOption>
- <configOption name="ice_support" default="no">
- <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
- </configOption>
- <configOption name="identify_by">
- <synopsis>Way(s) for the endpoint to be identified</synopsis>
- <description>
- <para>Endpoints and AORs can be identified in multiple ways. This
- option is a comma separated list of methods the endpoint can be
- identified.
- </para>
- <note><para>
- This option controls both how an endpoint is matched for incoming
- traffic and also how an AOR is determined if a registration
- occurs. You must list at least one method that also matches for
- AORs or the registration will fail.
- </para></note>
- <enumlist>
- <enum name="username">
- <para>Matches the endpoint or AOR ID based on the username
- and domain in the From header (or To header for AORs). If
- an exact match on both username and domain/realm fails, the
- match is retried with just the username.
- </para>
- </enum>
- <enum name="auth_username">
- <para>Matches the endpoint or AOR ID based on the username
- and realm in the Authentication header. If an exact match
- on both username and domain/realm fails, the match is
- retried with just the username.
- </para>
- <note><para>This method of identification has some security
- considerations because an Authentication header is not
- present on the first message of a dialog when digest
- authentication is used. The client can't generate it until
- the server sends the challenge in a 401 response. Since
- Asterisk normally sends a security event when an incoming
- request can't be matched to an endpoint, using this method
- requires that the security event be deferred until a request
- is received with the Authentication header and only
- generated if the username doesn't result in a match. This
- may result in a delay before an attack is recognized. You
- can control how many unmatched requests are received from
- a single ip address before a security event is generated
- using the <literal>unidentified_request</literal>
- parameters in the "global" configuration object.
- </para></note>
- </enum>
- <enum name="ip">
- <para>Matches the endpoint based on the source IP address.
- </para>
- <para>This method of identification is not configured here
- but simply allowed by this configuration option. See the
- documentation for the <literal>identify</literal>
- configuration section for more details on this method of
- endpoint identification.
- </para>
- </enum>
- <enum name="header">
- <para>Matches the endpoint based on a configured SIP header
- value.
- </para>
- <para>This method of identification is not configured here
- but simply allowed by this configuration option. See the
- documentation for the <literal>identify</literal>
- configuration section for more details on this method of
- endpoint identification.
- </para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="redirect_method">
- <synopsis>How redirects received from an endpoint are handled</synopsis>
- <description><para>
- When a redirect is received from an endpoint there are multiple ways it can be handled.
- If this option is set to <literal>user</literal> the user portion of the redirect target
- is treated as an extension within the dialplan and dialed using a Local channel. If this option
- is set to <literal>uri_core</literal> the target URI is returned to the dialing application
- which dials it using the PJSIP channel driver and endpoint originally used. If this option is
- set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
- to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
- and also supporting multiple potential redirect targets. The con is that since redirection occurs
- within chan_pjsip redirecting information is not forwarded and redirection can not be
- prevented.
- </para>
- <enumlist>
- <enum name="user" />
- <enum name="uri_core" />
- <enum name="uri_pjsip" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="mailboxes">
- <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
- <description><para>
- Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
- changes happen for any of the specified mailboxes. More than one mailbox can be
- specified with a comma-delimited string. app_voicemail mailboxes must be specified
- as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
- external sources, such as through the res_external_mwi module, you must specify
- strings supported by the external system.
- </para><para>
- For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
- configuration.
- </para></description>
- </configOption>
- <configOption name="mwi_subscribe_replaces_unsolicited">
- <synopsis>An MWI subscribe will replace sending unsolicited NOTIFYs</synopsis>
- </configOption>
- <configOption name="voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
- </configOption>
- <configOption name="moh_suggest" default="default">
- <synopsis>Default Music On Hold class</synopsis>
- </configOption>
- <configOption name="outbound_auth">
- <synopsis>Authentication object(s) used for outbound requests</synopsis>
- <description><para>
- This is a comma-delimited list of <replaceable>auth</replaceable>
- sections defined in <filename>pjsip.conf</filename> used to respond
- to outbound connection authentication challenges.</para>
- <note><para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses. See the auth realm description for details.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Full SIP URI of the outbound proxy used to send requests</synopsis>
- </configOption>
- <configOption name="rewrite_contact">
- <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
- <description><para>
- On inbound SIP messages from this endpoint, the Contact header or an
- appropriate Record-Route header will be changed to have the source IP
- address and port. This option does not affect outbound messages sent to
- this endpoint. This option helps servers communicate with endpoints
- that are behind NATs. This option also helps reuse reliable transport
- connections such as TCP and TLS.
- </para></description>
- </configOption>
- <configOption name="rtp_ipv6" default="no">
- <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
- </configOption>
- <configOption name="rtp_symmetric" default="no">
- <synopsis>Enforce that RTP must be symmetric</synopsis>
- </configOption>
- <configOption name="send_diversion" default="yes">
- <synopsis>Send the Diversion header, conveying the diversion
- information to the called user agent</synopsis>
- </configOption>
- <configOption name="send_pai" default="no">
- <synopsis>Send the P-Asserted-Identity header</synopsis>
- </configOption>
- <configOption name="send_rpid" default="no">
- <synopsis>Send the Remote-Party-ID header</synopsis>
- </configOption>
- <configOption name="rpid_immediate" default="no">
- <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
- <description>
- <para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
- or <emphasis>183 Progress</emphasis> response messages to the
- caller if the connected line information is updated before
- the call is answered. This can send a <emphasis>180 Ringing</emphasis>
- response before the call has even reached the far end. The
- caller can start hearing ringback before the far end even gets
- the call. Many phones tend to grab the first connected line
- information and refuse to update the display if it changes. The
- first information is not likely to be correct if the call
- goes to an endpoint not under the control of this Asterisk
- box.</para>
- <para>When disabled, a connected line update must wait for
- another reason to send a message with the connected line
- information to the caller before the call is answered. You can
- trigger the sending of the information by using an appropriate
- dialplan application such as <emphasis>Ringing</emphasis>.</para>
- </description>
- </configOption>
- <configOption name="timers_min_se" default="90">
- <synopsis>Minimum session timers expiration period</synopsis>
- <description><para>
- Minimum session timer expiration period. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="timers" default="yes">
- <synopsis>Session timers for SIP packets</synopsis>
- <description>
- <enumlist>
- <enum name="no" />
- <enum name="yes" />
- <enum name="required" />
- <enum name="always" />
- <enum name="forced"><para>Alias of always</para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="timers_sess_expires" default="1800">
- <synopsis>Maximum session timer expiration period</synopsis>
- <description><para>
- Maximum session timer expiration period. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="transport">
- <synopsis>Explicit transport configuration to use</synopsis>
- <description>
- <para>This will <emphasis>force</emphasis> the endpoint to use the
- specified transport configuration to send SIP messages. You need
- to already know what kind of transport (UDP/TCP/IPv4/etc) the
- endpoint device will use.
- </para>
- <note><para>Not specifying a transport will select the first
- configured transport in <filename>pjsip.conf</filename> which is
- compatible with the URI we are trying to contact.
- </para></note>
- <warning><para>Transport configuration is not affected by reloads. In order to
- change transports, a full Asterisk restart is required</para></warning>
- </description>
- </configOption>
- <configOption name="trust_id_inbound" default="no">
- <synopsis>Accept identification information received from this endpoint</synopsis>
- <description><para>This option determines whether Asterisk will accept
- identification from the endpoint from headers such as P-Asserted-Identity
- or Remote-Party-ID header. This option applies both to calls originating from the
- endpoint and calls originating from Asterisk. If <literal>no</literal>, the
- configured Caller-ID from pjsip.conf will always be used as the identity for
- the endpoint.</para></description>
- </configOption>
- <configOption name="trust_id_outbound" default="no">
- <synopsis>Send private identification details to the endpoint.</synopsis>
- <description><para>This option determines whether res_pjsip will send private
- identification information to the endpoint. If <literal>no</literal>,
- private Caller-ID information will not be forwarded to the endpoint.
- "Private" in this case refers to any method of restricting identification.
- Example: setting <replaceable>callerid_privacy</replaceable> to any
- <literal>prohib</literal> variation.
- Example: If <replaceable>trust_id_inbound</replaceable> is set to
- <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
- header in a SIP request or response would indicate the identification
- provided in the request is private.</para></description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'endpoint'.</synopsis>
- </configOption>
- <configOption name="use_ptime" default="no">
- <synopsis>Use Endpoint's requested packetization interval</synopsis>
- </configOption>
- <configOption name="use_avpf" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
- endpoint.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
- profile for all media offers on outbound calls and media updates and will
- decline media offers not using the AVPF or SAVPF profile.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
- profile for all media offers on outbound calls and media updates, and will
- decline media offers not using the AVP or SAVP profile.
- </para></description>
- </configOption>
- <configOption name="force_avp" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of AVP,
- regardless of the RTP profile in use for this endpoint.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the AVP, AVPF, SAVP, or
- SAVPF RTP profile for all media offers on outbound calls and media updates including
- those for DTLS-SRTP streams.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
- depending on configuration.
- </para></description>
- </configOption>
- <configOption name="media_use_received_transport" default="no">
- <synopsis>Determines whether res_pjsip will use the media transport received in the
- offer SDP in the corresponding answer SDP.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the received media transport.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
- depending on configuration.
- </para></description>
- </configOption>
- <configOption name="media_encryption" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
- for this endpoint.</synopsis>
- <description>
- <enumlist>
- <enum name="no"><para>
- res_pjsip will offer no encryption and allow no encryption to be setup.
- </para></enum>
- <enum name="sdes"><para>
- res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
- transport should be used in conjunction with this option to prevent
- exposure of media encryption keys.
- </para></enum>
- <enum name="dtls"><para>
- res_pjsip will offer DTLS-SRTP setup.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="media_encryption_optimistic" default="no">
- <synopsis>Determines whether encryption should be used if possible but does not terminate the
- session if not achieved.</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>sdes</literal> or <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="g726_non_standard" default="no">
- <synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
- <description><para>
- When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
- packing order instead of what is recommended by RFC3551. Since this essentially
- replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
- specified in the endpoint's allowed codec list.
- </para></description>
- </configOption>
- <configOption name="inband_progress" default="no">
- <synopsis>Determines whether chan_pjsip will indicate ringing using inband
- progress.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
- when told to indicate ringing and will immediately start sending ringing
- as audio.
- </para><para>
- If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
- to indicate ringing and will NOT send it as audio.
- </para></description>
- </configOption>
- <configOption name="call_group">
- <synopsis>The numeric pickup groups for a channel.</synopsis>
- <description><para>
- Can be set to a comma separated list of numbers or ranges between the values
- of 0-63 (maximum of 64 groups).
- </para></description>
- </configOption>
- <configOption name="pickup_group">
- <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
- <description><para>
- Can be set to a comma separated list of numbers or ranges between the values
- of 0-63 (maximum of 64 groups).
- </para></description>
- </configOption>
- <configOption name="named_call_group">
- <synopsis>The named pickup groups for a channel.</synopsis>
- <description><para>
- Can be set to a comma separated list of case sensitive strings limited by
- supported line length.
- </para></description>
- </configOption>
- <configOption name="named_pickup_group">
- <synopsis>The named pickup groups that a channel can pickup.</synopsis>
- <description><para>
- Can be set to a comma separated list of case sensitive strings limited by
- supported line length.
- </para></description>
- </configOption>
- <configOption name="device_state_busy_at" default="0">
- <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
- <description><para>
- When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
- PJSIP channel driver will return busy as the device state instead of in use.
- </para></description>
- </configOption>
- <configOption name="t38_udptl" default="no">
- <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
- <description><para>
- If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
- and relayed.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_ec" default="none">
- <synopsis>T.38 UDPTL error correction method</synopsis>
- <description>
- <enumlist>
- <enum name="none"><para>
- No error correction should be used.
- </para></enum>
- <enum name="fec"><para>
- Forward error correction should be used.
- </para></enum>
- <enum name="redundancy"><para>
- Redundancy error correction should be used.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="t38_udptl_maxdatagram" default="0">
- <synopsis>T.38 UDPTL maximum datagram size</synopsis>
- <description><para>
- This option can be set to override the maximum datagram of a remote endpoint for broken
- endpoints.
- </para></description>
- </configOption>
- <configOption name="fax_detect" default="no">
- <synopsis>Whether CNG tone detection is enabled</synopsis>
- <description><para>
- This option can be set to send the session to the fax extension when a CNG tone is
- detected.
- </para></description>
- </configOption>
- <configOption name="fax_detect_timeout">
- <synopsis>How long into a call before fax_detect is disabled for the call</synopsis>
- <description><para>
- The option determines how many seconds into a call before the
- fax_detect option is disabled for the call. Setting the value
- to zero disables the timeout.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_nat" default="no">
- <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
- <description><para>
- When enabled the UDPTL stack will send UDPTL packets to the source address of
- received packets.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_ipv6" default="no">
- <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
- <description><para>
- When enabled the UDPTL stack will use IPv6.
- </para></description>
- </configOption>
- <configOption name="tone_zone">
- <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
- </configOption>
- <configOption name="language">
- <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
- </configOption>
- <configOption name="one_touch_recording" default="no">
- <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
- <see-also>
- <ref type="configOption">record_on_feature</ref>
- <ref type="configOption">record_off_feature</ref>
- </see-also>
- </configOption>
- <configOption name="record_on_feature" default="automixmon">
- <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
- <description>
- <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
- feature will be enabled for the channel. The feature designated here can be any built-in
- or dynamic feature defined in features.conf.</para>
- <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
- </description>
- <see-also>
- <ref type="configOption">one_touch_recording</ref>
- <ref type="configOption">record_off_feature</ref>
- </see-also>
- </configOption>
- <configOption name="record_off_feature" default="automixmon">
- <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
- <description>
- <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
- feature will be enabled for the channel. The feature designated here can be any built-in
- or dynamic feature defined in features.conf.</para>
- <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
- </description>
- <see-also>
- <ref type="configOption">one_touch_recording</ref>
- <ref type="configOption">record_on_feature</ref>
- </see-also>
- </configOption>
- <configOption name="rtp_engine" default="asterisk">
- <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
- </configOption>
- <configOption name="allow_transfer" default="yes">
- <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
- </configOption>
- <configOption name="user_eq_phone" default="no">
- <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
- </configOption>
- <configOption name="sdp_owner" default="-">
- <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
- </configOption>
- <configOption name="sdp_session" default="Asterisk">
- <synopsis>String used for the SDP session (s=) line.</synopsis>
- </configOption>
- <configOption name="tos_audio">
- <synopsis>DSCP TOS bits for audio streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="tos_video">
- <synopsis>DSCP TOS bits for video streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="cos_audio">
- <synopsis>Priority for audio streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="cos_video">
- <synopsis>Priority for video streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="allow_subscribe" default="yes">
- <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
- </configOption>
- <configOption name="sub_min_expiry" default="60">
- <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
- </configOption>
- <configOption name="from_user">
- <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
- </configOption>
- <configOption name="mwi_from_user">
- <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
- </configOption>
- <configOption name="from_domain">
- <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
- </configOption>
- <configOption name="dtls_verify">
- <synopsis>Verify that the provided peer certificate is valid</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_rekey">
- <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para><para>
- If this is not set or the value provided is 0 rekeying will be disabled.
- </para></description>
- </configOption>
- <configOption name="dtls_cert_file">
- <synopsis>Path to certificate file to present to peer</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_private_key">
- <synopsis>Path to private key for certificate file</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_cipher">
- <synopsis>Cipher to use for DTLS negotiation</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <para>Many options for acceptable ciphers. See link for more:</para>
- <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
- </para></description>
- </configOption>
- <configOption name="dtls_ca_file">
- <synopsis>Path to certificate authority certificate</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_ca_path">
- <synopsis>Path to a directory containing certificate authority certificates</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_setup">
- <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
- <description>
- <para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <enumlist>
- <enum name="active"><para>
- res_pjsip will make a connection to the peer.
- </para></enum>
- <enum name="passive"><para>
- res_pjsip will accept connections from the peer.
- </para></enum>
- <enum name="actpass"><para>
- res_pjsip will offer and accept connections from the peer.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="dtls_fingerprint">
- <synopsis>Type of hash to use for the DTLS fingerprint in the SDP.</synopsis>
- <description>
- <para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <enumlist>
- <enum name="SHA-256"></enum>
- <enum name="SHA-1"></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="srtp_tag_32">
- <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>sdes</literal> or <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="set_var">
- <synopsis>Variable set on a channel involving the endpoint.</synopsis>
- <description><para>
- When a new channel is created using the endpoint set the specified
- variable(s) on that channel. For multiple channel variables specify
- multiple 'set_var'(s).
- </para></description>
- </configOption>
- <configOption name="message_context">
- <synopsis>Context to route incoming MESSAGE requests to.</synopsis>
- <description><para>
- If specified, incoming MESSAGE requests will be routed to the indicated
- dialplan context. If no <replaceable>message_context</replaceable> is
- specified, then the <replaceable>context</replaceable> setting is used.
- </para></description>
- </configOption>
- <configOption name="accountcode">
- <synopsis>An accountcode to set automatically on any channels created for this endpoint.</synopsis>
- <description><para>
- If specified, any channel created for this endpoint will automatically
- have this accountcode set on it.
- </para></description>
- </configOption>
- <configOption name="rtp_keepalive">
- <synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
- <description><para>
- At the specified interval, Asterisk will send an RTP comfort noise frame. This may
- be useful for situations where Asterisk is behind a NAT or firewall and must keep a
- hole open in order to allow for media to arrive at Asterisk.
- </para></description>
- </configOption>
- <configOption name="rtp_timeout" default="0">
- <synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
- <description><para>
- This option configures the number of seconds without RTP (while off hold) before
- considering a channel as dead. When the number of seconds is reached the underlying
- channel is hung up. By default this option is set to 0, which means do not check.
- </para></description>
- </configOption>
- <configOption name="rtp_timeout_hold" default="0">
- <synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
- <description><para>
- This option configures the number of seconds without RTP (while on hold) before
- considering a channel as dead. When the number of seconds is reached the underlying
- channel is hung up. By default this option is set to 0, which means do not check.
- </para></description>
- </configOption>
- <configOption name="acl">
- <synopsis>List of IP ACL section names in acl.conf</synopsis>
- <description><para>
- This matches sections configured in <literal>acl.conf</literal>. The value is
- defined as a list of comma-delimited section names.
- </para></description>
- </configOption>
- <configOption name="deny">
- <synopsis>List of IP addresses to deny access from</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="permit">
- <synopsis>List of IP addresses to permit access from</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="contact_acl">
- <synopsis>List of Contact ACL section names in acl.conf</synopsis>
- <description><para>
- This matches sections configured in <literal>acl.conf</literal>. The value is
- defined as a list of comma-delimited section names.
- </para></description>
- </configOption>
- <configOption name="contact_deny">
- <synopsis>List of Contact header addresses to deny</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="contact_permit">
- <synopsis>List of Contact header addresses to permit</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="subscribe_context">
- <synopsis>Context for incoming MESSAGE requests.</synopsis>
- <description><para>
- If specified, incoming SUBSCRIBE requests will be searched for the matching
- extension in the indicated context.
- If no <replaceable>subscribe_context</replaceable> is specified,
- then the <replaceable>context</replaceable> setting is used.
- </para></description>
- </configOption>
- <configOption name="contact_user" default="">
- <synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
- <description><para>
- On outbound requests, force the user portion of the Contact header to this value.
- </para></description>
- </configOption>
- <configOption name="asymmetric_rtp_codec" default="no">
- <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
- <description><para>
- When set to "yes" the codec in use for sending will be allowed to differ from
- that of the received one. PJSIP will not automatically switch the sending one
- to the receiving one.
- </para></description>
- </configOption>
- <configOption name="rtcp_mux" default="no">
- <synopsis>Enable RFC 5761 RTCP multiplexing on the RTP port</synopsis>
- <description><para>
- With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux"
- attribute on all media streams. This will result in RTP and RTCP being sent and received
- on the same port. This shifts the demultiplexing logic to the application rather than
- the transport layer. This option is useful when interoperating with WebRTC endpoints
- since they mandate this option's use.
- </para></description>
- </configOption>
- <configOption name="refer_blind_progress" default="yes">
- <synopsis>Whether to notifies all the progress details on blind transfer</synopsis>
- <description><para>
- Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK"
- after REFER has been accepted. If set to <literal>no</literal> then asterisk
- will not send the progress details, but immediately will send "200 OK".
- </para></description>
- </configOption>
- <configOption name="notify_early_inuse_ringing" default="no">
- <synopsis>Whether to notifies dialog-info 'early' on InUse&Ringing state</synopsis>
- <description><para>
- Control whether dialog-info subscriptions get 'early' state
- on Ringing when already INUSE.
- </para></description>
- </configOption>
- <configOption name="incoming_mwi_mailbox">
- <synopsis>Mailbox name to use when incoming MWI NOTIFYs are received</synopsis>
- <description><para>
- If an MWI NOTIFY is received <emphasis>from</emphasis> this endpoint,
- this mailbox will be used when notifying other modules of MWI status
- changes. If not set, incoming MWI NOTIFYs are ignored.
- </para></description>
- </configOption>
- <configOption name="follow_early_media_fork">
- <synopsis>Follow SDP forked media when To tag is different</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on subsequent 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is different than that on the previous
- one, follow it. This usually happens when the INVITE is forked to multiple
- UASs and more than one sends an SDP answer.
- </para>
- <note><para>
- This option must also be enabled in the <literal>system</literal>
- section for it to take effect here.
- </para></note>
- </description>
- </configOption>
- <configOption name="accept_multiple_sdp_answers" default="no">
- <synopsis>Accept multiple SDP answers on non-100rel responses</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on non-100rel 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is the same as that on the previous one,
- process the updated SDP. This can happen when the UAS needs to change ports
- for some reason such as using a separate port for custom ringback.
- </para>
- <note><para>
- This option must also be enabled in the <literal>system</literal>
- section for it to take effect here.
- </para></note>
- </description>
- </configOption>
- <configOption name="suppress_q850_reason_headers" default="no">
- <synopsis>Suppress Q.850 Reason headers for this endpoint</synopsis>
- <description><para>
- Some devices can't accept multiple Reason headers and get confused
- when both 'SIP' and 'Q.850' Reason headers are received. This
- option allows the 'Q.850' Reason header to be suppressed.</para>
- </description>
- </configOption>
- </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>Plain text password used for authentication.</synopsis>
- <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
- </configOption>
- <configOption name="realm">
- <synopsis>SIP realm for endpoint</synopsis>
- <description><para>
- The treatment of this value depends upon how the authentication
- object is used.
- </para><para>
- When used as an inbound authentication object, the realm is sent
- as part of the challenge so the peer can know which key to use
- when responding. An empty value will use the
- <replaceable>global</replaceable> section's
- <literal>default_realm</literal> value when issuing a challenge.
- </para><para>
- When used as an outbound authentication object, the realm is
- matched with the received challenge realm to determine which
- authentication object to use when responding to the challenge. An
- empty value matches any challenging realm when determining
- which authentication object matches a received challenge.
- </para>
- <note><para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses.</para></note>
- </description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be 'auth'</synopsis>
- </configOption>
- <configOption name="username">
- <synopsis>Username to use for account</synopsis>
- </configOption>
- </configObject>
- <configObject name="domain_alias">
- <synopsis>Domain Alias</synopsis>
- <description><para>
- Signifies that a domain is an alias. If the domain on a session is
- not found to match an AoR then this object is used to see if we have
- an alias for the AoR to which the endpoint is binding. This objects
- name as defined in configuration should be the domain alias and a
- config option is provided to specify the domain to be aliased.
- </para></description>
- <configOption name="type">
- <synopsis>Must be of type 'domain_alias'.</synopsis>
- </configOption>
- <configOption name="domain">
- <synopsis>Domain to be aliased</synopsis>
- </configOption>
- </configObject>
- <configObject name="transport">
- <synopsis>SIP Transport</synopsis>
- <description><para>
- <emphasis>Transports</emphasis>
- </para>
- <para>There are different transports and protocol derivatives
- supported by <literal>res_pjsip</literal>. They are in order of
- preference: UDP, TCP, and WebSocket (WS).</para>
- <note><para>Changes to transport configuration in pjsip.conf will only be
- effected on a complete restart of Asterisk. A module reload
- will not suffice.</para></note>
- </description>
- <configOption name="async_operations" default="1">
- <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
- </configOption>
- <configOption name="bind">
- <synopsis>IP Address and optional port to bind to for this transport</synopsis>
- </configOption>
- <configOption name="ca_list_file">
- <synopsis>File containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="ca_list_path">
- <synopsis>Path to directory containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="cert_file">
- <synopsis>Certificate file for endpoint (TLS ONLY, not WSS)</synopsis>
- <description><para>
- A path to a .crt or .pem file can be provided. However, only
- the certificate is read from the file, not the private key.
- The <literal>priv_key_file</literal> option must supply a
- matching key file.
- </para></description>
- </configOption>
- <configOption name="cipher">
- <synopsis>Preferred cryptography cipher names (TLS ONLY, not WSS)</synopsis>
- <description>
- <para>Comma separated list of cipher names or numeric equivalents.
- Numeric equivalents can be either decimal or hexadecimal (0xX).
- </para>
- <para>There are many cipher names. Use the CLI command
- <literal>pjsip list ciphers</literal> to see a list of cipher
- names available for your installation. See link for more:</para>
- <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES
- </para>
- </description>
- </configOption>
- <configOption name="domain">
- <synopsis>Domain the transport comes from</synopsis>
- </configOption>
- <configOption name="external_media_address">
- <synopsis>External IP address to use in RTP handling</synopsis>
- <description><para>
- When a request or response is sent out, if the destination of the
- message is outside the IP network defined in the option <literal>localnet</literal>,
- and the media address in the SDP is within the localnet network, then the
- media address in the SDP will be rewritten to the value defined for
- <literal>external_media_address</literal>.
- </para></description>
- </configOption>
- <configOption name="external_signaling_address">
- <synopsis>External address for SIP signalling</synopsis>
- </configOption>
- <configOption name="external_signaling_port" default="0">
- <synopsis>External port for SIP signalling</synopsis>
- </configOption>
- <configOption name="method">
- <synopsis>Method of SSL transport (TLS ONLY, not WSS)</synopsis>
- <description>
- <enumlist>
- <enum name="default">
- <para>The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.</para>
- </enum>
- <enum name="unspecified">
- <para>This option is equivalent to setting 'default'</para>
- </enum>
- <enum name="tlsv1" />
- <enum name="tlsv1_1" />
- <enum name="tlsv1_2" />
- <enum name="sslv2" />
- <enum name="sslv3" />
- <enum name="sslv23" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="local_net">
- <synopsis>Network to consider local (used for NAT purposes).</synopsis>
- <description><para>This must be in CIDR or dotted decimal format with the IP
- and mask separated with a slash ('/').</para></description>
- </configOption>
- <configOption name="password">
- <synopsis>Password required for transport</synopsis>
- </configOption>
- <configOption name="priv_key_file">
- <synopsis>Private key file (TLS ONLY, not WSS)</synopsis>
- </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, not WSS)</synopsis>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'transport'.</synopsis>
- </configOption>
- <configOption name="verify_client" default="false">
- <synopsis>Require verification of client certificate (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="verify_server" default="false">
- <synopsis>Require verification of server certificate (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="tos" default="false">
- <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
- <description>
- <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
- for more information on this parameter.</para>
- <note><para>This option does not apply to the <replaceable>ws</replaceable>
- or the <replaceable>wss</replaceable> protocols.</para></note>
- </description>
- </configOption>
- <configOption name="cos" default="false">
- <synopsis>Enable COS for the signalling sent over this transport</synopsis>
- <description>
- <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
- for more information on this parameter.</para>
- <note><para>This option does not apply to the <replaceable>ws</replaceable>
- or the <replaceable>wss</replaceable> protocols.</para></note>
- </description>
- </configOption>
- <configOption name="websocket_write_timeout">
- <synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
- <description>
- <para>If a websocket connection accepts input slowly, the timeout
- for writes to it can be increased to keep it from being disconnected.
- Value is in milliseconds; default is 100 ms.</para>
- </description>
- </configOption>
- <configOption name="allow_reload" default="no">
- <synopsis>Allow this transport to be reloaded.</synopsis>
- <description>
- <para>Allow this transport to be reloaded when res_pjsip is reloaded.
- This option defaults to "no" because reloading a transport may disrupt
- in-progress calls.</para>
- </description>
- </configOption>
- <configOption name="symmetric_transport" default="no">
- <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis>
- <description>
- <para>When a request from a dynamic contact
- comes in on a transport with this option set to 'yes',
- the transport name will be saved and used for subsequent
- outgoing requests like OPTIONS, NOTIFY and INVITE. It's
- saved as a contact uri parameter named 'x-ast-txp' and will
- display with the contact uri in CLI, AMI, and ARI output.
- On the outgoing request, if a transport wasn't explicitly
- set on the endpoint AND the request URI is not a hostname,
- the saved transport will be used and the 'x-ast-txp'
- parameter stripped from the outgoing packet.
- </para>
- </description>
- </configOption>
- </configObject>
- <configObject name="contact">
- <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
- <description><para>
- Contacts are a way to hide SIP URIs from the dialplan directly.
- They are also used to make a group of contactable parties when
- in use with <literal>AoR</literal> lists.
- </para></description>
- <configOption name="type">
- <synopsis>Must be of type 'contact'.</synopsis>
- </configOption>
- <configOption name="uri">
- <synopsis>SIP URI to contact peer</synopsis>
- </configOption>
- <configOption name="expiration_time">
- <synopsis>Time to keep alive a contact</synopsis>
- <description><para>
- Time to keep alive a contact. String style specification.
- </para></description>
- </configOption>
- <configOption name="qualify_frequency" default="0">
- <synopsis>Interval at which to qualify a contact</synopsis>
- <description><para>
- Interval between attempts to qualify the contact for reachability.
- If <literal>0</literal> never qualify. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="qualify_timeout" default="3.0">
- <synopsis>Timeout for qualify</synopsis>
- <description><para>
- If the contact doesn't respond to the OPTIONS request before the timeout,
- the contact is marked unavailable.
- If <literal>0</literal> no timeout. Time in fractional seconds.
- </para></description>
- </configOption>
- <configOption name="authenticate_qualify">
- <synopsis>Authenticates a qualify challenge response if needed</synopsis>
- <description>
- <para>If true and a qualify request receives a challenge response then
- authentication is attempted before declaring the contact available.
- </para>
- <note><para>This option does nothing as we will always complete
- the challenge response authentication if the qualify request is
- challenged.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
- <description><para>
- If set the provided URI will be used as the outbound proxy when an
- OPTIONS request is sent to a contact for qualify purposes.
- </para></description>
- </configOption>
- <configOption name="path">
- <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
- </configOption>
- <configOption name="user_agent">
- <synopsis>User-Agent header from registration.</synopsis>
- <description><para>
- The User-Agent is automatically stored based on data present in incoming SIP
- REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="endpoint">
- <synopsis>Endpoint name</synopsis>
- <description><para>
- The name of the endpoint this contact belongs to
- </para></description>
- </configOption>
- <configOption name="reg_server">
- <synopsis>Asterisk Server name</synopsis>
- <description><para>
- Asterisk Server name on which SIP endpoint registered.
- </para></description>
- </configOption>
- <configOption name="via_addr">
- <synopsis>IP-address of the last Via header from registration.</synopsis>
- <description><para>
- The last Via header should contain the address of UA which sent the request.
- The IP-address of the last Via header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="via_port">
- <synopsis>IP-port of the last Via header from registration.</synopsis>
- <description><para>
- The IP-port of the last Via header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="call_id">
- <synopsis>Call-ID header from registration.</synopsis>
- <description><para>
- The Call-ID header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="prune_on_boot">
- <synopsis>A contact that cannot survive a restart/boot.</synopsis>
- <description><para>
- The option is set if the incoming SIP REGISTER contact is rewritten
- on a reliable transport and is not intended to be configured manually.
- </para></description>
- </configOption>
- </configObject>
- <configObject name="aor">
- <synopsis>The configuration for a location of an endpoint</synopsis>
- <description><para>
- An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
- AoRs are specified, an endpoint will not be reachable by Asterisk.
- Beyond that, an AoR has other uses within Asterisk, such as inbound
- registration.
- </para><para>
- An <literal>AoR</literal> is a way to allow dialing a group
- of <literal>Contacts</literal> that all use the same
- <literal>endpoint</literal> for calls.
- </para><para>
- This can be used as another way of grouping a list of contacts to dial
- rather than specifying them each directly when dialing via the dialplan.
- This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
- </para><para>
- Registrations: For Asterisk to match an inbound registration to an endpoint,
- the AoR object name must match the user portion of the SIP URI in the "To:"
- header of the inbound SIP registration. That will usually be equivalent
- to the "user name" set in your hard or soft phones configuration.
- </para></description>
- <configOption name="contact">
- <synopsis>Permanent contacts assigned to AoR</synopsis>
- <description><para>
- Contacts specified will be called whenever referenced
- by <literal>chan_pjsip</literal>.
- </para><para>
- Use a separate "contact=" entry for each contact required. Contacts
- are specified using a SIP URI.
- </para></description>
- </configOption>
- <configOption name="default_expiration" default="3600">
- <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
- </configOption>
- <configOption name="mailboxes">
- <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
- <description><para>This option applies when an external entity subscribes to an AoR
- for Message Waiting Indications. The mailboxes specified will be subscribed to.
- More than one mailbox can be specified with a comma-delimited string.
- app_voicemail mailboxes must be specified as mailbox@context;
- for example: mailboxes=6001@default. For mailboxes provided by external sources,
- such as through the res_external_mwi module, you must specify strings supported by
- the external system.
- </para><para>
- For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
- endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
- </para></description>
- </configOption>
- <configOption name="voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
- </configOption>
- <configOption name="maximum_expiration" default="7200">
- <synopsis>Maximum time to keep an AoR</synopsis>
- <description><para>
- Maximum time to keep a peer with explicit expiration. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="max_contacts" default="0">
- <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
- <description><para>
- Maximum number of contacts that can associate with this AoR. This value does
- not affect the number of contacts that can be added with the "contact" option.
- It only limits contacts added through external interaction, such as
- registration.
- </para>
- <note><para>The <replaceable>rewrite_contact</replaceable> option
- registers the source address as the contact address to help with
- NAT and reusing connection oriented transports such as TCP and
- TLS. Unfortunately, refreshing a registration may register a
- different contact address and exceed
- <replaceable>max_contacts</replaceable>. The
- <replaceable>remove_existing</replaceable> option can help by
- removing the soonest to expire contact(s) over
- <replaceable>max_contacts</replaceable> which is likely the
- old <replaceable>rewrite_contact</replaceable> contact source
- address being refreshed.
- </para></note>
- <note><para>This should be set to <literal>1</literal> and
- <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
- wish to stick with the older <literal>chan_sip</literal> behaviour.
- </para></note>
- </description>
- </configOption>
- <configOption name="minimum_expiration" default="60">
- <synopsis>Minimum keep alive time for an AoR</synopsis>
- <description><para>
- Minimum time to keep a peer with an explicit expiration. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="remove_existing" default="no">
- <synopsis>Determines whether new contacts replace existing ones.</synopsis>
- <description><para>
- On receiving a new registration to the AoR should it remove enough
- existing contacts not added or updated by the registration to
- satisfy <replaceable>max_contacts</replaceable>? Any removed
- contacts will expire the soonest.
- </para>
- <note><para>The <replaceable>rewrite_contact</replaceable> option
- registers the source address as the contact address to help with
- NAT and reusing connection oriented transports such as TCP and
- TLS. Unfortunately, refreshing a registration may register a
- different contact address and exceed
- <replaceable>max_contacts</replaceable>. The
- <replaceable>remove_existing</replaceable> option can help by
- removing the soonest to expire contact(s) over
- <replaceable>max_contacts</replaceable> which is likely the
- old <replaceable>rewrite_contact</replaceable> contact source
- address being refreshed.
- </para></note>
- <note><para>This should be set to <literal>yes</literal> and
- <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
- wish to stick with the older <literal>chan_sip</literal> behaviour.
- </para></note>
- </description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'aor'.</synopsis>
- </configOption>
- <configOption name="qualify_frequency" default="0">
- <synopsis>Interval at which to qualify an AoR</synopsis>
- <description><para>
- Interval between attempts to qualify the AoR for reachability.
- If <literal>0</literal> never qualify. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="qualify_timeout" default="3.0">
- <synopsis>Timeout for qualify</synopsis>
- <description><para>
- If the contact doesn't respond to the OPTIONS request before the timeout,
- the contact is marked unavailable.
- If <literal>0</literal> no timeout. Time in fractional seconds.
- </para></description>
- </configOption>
- <configOption name="authenticate_qualify">
- <synopsis>Authenticates a qualify challenge response if needed</synopsis>
- <description>
- <para>If true and a qualify request receives a challenge response then
- authentication is attempted before declaring the contact available.
- </para>
- <note><para>This option does nothing as we will always complete
- the challenge response authentication if the qualify request is
- challenged.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
- <description><para>
- If set the provided URI will be used as the outbound proxy when an
- OPTIONS request is sent to a contact for qualify purposes.
- </para></description>
- </configOption>
- <configOption name="support_path">
- <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
- <description><para>
- When this option is enabled, the Path headers in register requests will be saved
- and its contents will be used in Route headers for outbound out-of-dialog requests
- and in Path headers for outbound 200 responses. Path support will also be indicated
- in the Supported header.
- </para></description>
- </configOption>
- </configObject>
- <configObject name="system">
- <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
- <description><para>
- The settings in this section are global. In addition to being global, the values will
- not be re-evaluated when a reload is performed. This is because the values must be set
- before the SIP stack is initialized. The only way to reset these values is to either
- restart Asterisk, or unload res_pjsip.so and then load it again.
- </para></description>
- <configOption name="timer_t1" default="500">
- <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
- <description><para>
- Timer T1 is the base for determining how long to wait before retransmitting
- requests that receive no response when using an unreliable transport (e.g. UDP).
- For more information on this timer, see RFC 3261, Section 17.1.1.1.
- </para></description>
- </configOption>
- <configOption name="timer_b" default="32000">
- <synopsis>Set transaction timer B value (milliseconds).</synopsis>
- <description><para>
- Timer B determines the maximum amount of time to wait after sending an INVITE
- request before terminating the transaction. It is recommended that this be set
- to 64 * Timer T1, but it may be set higher if desired. For more information on
- this timer, see RFC 3261, Section 17.1.1.1.
- </para></description>
- </configOption>
- <configOption name="compact_headers" default="no">
- <synopsis>Use the short forms of common SIP header names.</synopsis>
- </configOption>
- <configOption name="threadpool_initial_size" default="0">
- <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
- </configOption>
- <configOption name="threadpool_auto_increment" default="5">
- <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
- </configOption>
- <configOption name="threadpool_idle_timeout" default="60">
- <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
- </configOption>
- <configOption name="threadpool_max_size" default="0">
- <synopsis>Maximum number of threads in the res_pjsip threadpool.
- A value of 0 indicates no maximum.</synopsis>
- </configOption>
- <configOption name="disable_tcp_switch" default="yes">
- <synopsis>Disable automatic switching from UDP to TCP transports.</synopsis>
- <description><para>
- Disable automatic switching from UDP to TCP transports if outgoing
- request is too large. See RFC 3261 section 18.1.1.
- </para></description>
- </configOption>
- <configOption name="follow_early_media_fork">
- <synopsis>Follow SDP forked media when To tag is different</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on subsequent 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is different than that on the previous
- one, follow it.
- </para>
- <note><para>
- This option must also be enabled on endpoints that require
- this functionality.
- </para></note>
- </description>
- </configOption>
- <configOption name="accept_multiple_sdp_answers">
- <synopsis>Follow SDP forked media when To tag is the same</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on non-100rel 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is the same as that on the previous one,
- process the updated SDP.
- </para>
- <note><para>
- This option must also be enabled on endpoints that require
- this functionality.
- </para></note>
- </description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'system' UNLESS the object name is 'system'.</synopsis>
- </configOption>
- </configObject>
- <configObject name="global">
- <synopsis>Options that apply globally to all SIP communications</synopsis>
- <description><para>
- The settings in this section are global. Unlike options in the <literal>system</literal>
- section, these options can be refreshed by performing a reload.
- </para></description>
- <configOption name="max_forwards" default="70">
- <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
- </configOption>
- <configOption name="keep_alive_interval" default="90">
- <synopsis>The interval (in seconds) to send keepalives to active connection-oriented transports.</synopsis>
- </configOption>
- <configOption name="contact_expiration_check_interval" default="30">
- <synopsis>The interval (in seconds) to check for expired contacts.</synopsis>
- </configOption>
- <configOption name="disable_multi_domain" default="no">
- <synopsis>Disable Multi Domain support</synopsis>
- <description><para>
- If disabled it can improve realtime performance by reducing the number of database requests.
- </para></description>
- </configOption>
- <configOption name="max_initial_qualify_time" default="0">
- <synopsis>The maximum amount of time from startup that qualifies should be attempted on all contacts.
- If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.</synopsis>
- </configOption>
- <configOption name="unidentified_request_period" default="5">
- <synopsis>The number of seconds over which to accumulate unidentified requests.</synopsis>
- <description><para>
- If <literal>unidentified_request_count</literal> unidentified requests are received
- during <literal>unidentified_request_period</literal>, a security event will be generated.
- </para></description>
- </configOption>
- <configOption name="unidentified_request_count" default="5">
- <synopsis>The number of unidentified requests from a single IP to allow.</synopsis>
- <description><para>
- If <literal>unidentified_request_count</literal> unidentified requests are received
- during <literal>unidentified_request_period</literal>, a security event will be generated.
- </para></description>
- </configOption>
- <configOption name="unidentified_request_prune_interval" default="30">
- <synopsis>The interval at which unidentified requests are older than
- twice the unidentified_request_period are pruned.</synopsis>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'global' UNLESS the object name is 'global'.</synopsis>
- </configOption>
- <configOption name="user_agent" default="Asterisk <Asterisk Version>">
- <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
- </configOption>
- <configOption name="regcontext" default="">
- <synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
- peer who registers or unregisters with us.</synopsis>
- </configOption>
- <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
- <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
- </configOption>
- <configOption name="default_voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor</synopsis>
- </configOption>
- <configOption name="debug" default="no">
- <synopsis>Enable/Disable SIP debug logging. Valid options include yes, no, or
- a host address</synopsis>
- </configOption>
- <configOption name="endpoint_identifier_order">
- <synopsis>The order by which endpoint identifiers are processed and checked.
- Identifier names are usually derived from and can be found in the endpoint
- identifier module itself (res_pjsip_endpoint_identifier_*).
- You can use the CLI command "pjsip show identifiers" to see the
- identifiers currently available.</synopsis>
- <description>
- <note><para>
- One of the identifiers is "auth_username" which matches on the username in
- an Authentication header. This method has some security considerations because an
- Authentication header is not present on the first message of a dialog when
- digest authentication is used. The client can't generate it until the server
- sends the challenge in a 401 response. Since Asterisk normally sends a security
- event when an incoming request can't be matched to an endpoint, using auth_username
- requires that the security event be deferred until a request is received with
- the Authentication header and only generated if the username doesn't result in a
- match. This may result in a delay before an attack is recognized. You can control
- how many unmatched requests are received from a single ip address before a security
- event is generated using the unidentified_request parameters.
- </para></note>
- </description>
- </configOption>
- <configOption name="default_from_user" default="asterisk">
- <synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
- set to this value if there is no better option (such as CallerID) to be
- used.</synopsis>
- </configOption>
- <configOption name="default_realm" default="asterisk">
- <synopsis>When Asterisk generates a challenge, the digest realm will be
- set to this value if there is no better option (such as auth/realm) to be
- used.</synopsis>
- </configOption>
- <configOption name="mwi_tps_queue_high" default="500">
- <synopsis>MWI taskprocessor high water alert trigger level.</synopsis>
- <description>
- <para>On a heavily loaded system you may need to adjust the
- taskprocessor queue limits. If any taskprocessor queue size
- reaches its high water level then pjsip will stop processing
- new requests until the alert is cleared. The alert clears
- when all alerting taskprocessor queues have dropped to their
- low water clear level.
- </para>
- </description>
- </configOption>
- <configOption name="mwi_tps_queue_low" default="-1">
- <synopsis>MWI taskprocessor low water clear alert level.</synopsis>
- <description>
- <para>On a heavily loaded system you may need to adjust the
- taskprocessor queue limits. If any taskprocessor queue size
- reaches its high water level then pjsip will stop processing
- new requests until the alert is cleared. The alert clears
- when all alerting taskprocessor queues have dropped to their
- low water clear level.
- </para>
- <note><para>Set to -1 for the low water level to be 90% of
- the high water level.</para></note>
- </description>
- </configOption>
- <configOption name="mwi_disable_initial_unsolicited" default="no">
- <synopsis>Enable/Disable sending unsolicited MWI to all endpoints on startup.</synopsis>
- <description>
- <para>When the initial unsolicited MWI notification are
- enabled on startup then the initial notifications
- get sent at startup. If you have a lot of endpoints
- (thousands) that use unsolicited MWI then you may
- want to consider disabling the initial startup
- notifications.
- </para>
- <para>When the initial unsolicited MWI notifications are
- disabled on startup then the notifications will start
- on the endpoint's next contact update.
- </para>
- </description>
- </configOption>
- <configOption name="ignore_uri_user_options">
- <synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
- <description>
- <para>If you have this option enabled and there are semicolons
- in the user field of a SIP URI then the field is truncated
- at the first semicolon. This effectively makes the semicolon
- a non-usable character for PJSIP endpoint names, extensions,
- and AORs. This can be useful for improving compatibility with
- an ITSP that likes to use user options for whatever reason.
- </para>
- <example title="Sample SIP URI">
- sip:1235557890;phone-context=national@x.x.x.x;user=phone
- </example>
- <example title="Sample SIP URI user field">
- 1235557890;phone-context=national
- </example>
- <example title="Sample SIP URI user field truncated">
- 1235557890
- </example>
- <note><para>The caller-id and redirecting number strings
- obtained from incoming SIP URI user fields are always truncated
- at the first semicolon.</para></note>
- </description>
- </configOption>
- <configOption name="use_callerid_contact" default="no">
- <synopsis>Place caller-id information into Contact header</synopsis>
- <description><para>
- This option will cause Asterisk to place caller-id information into
- generated Contact headers.</para>
- </description>
- </configOption>
- <configOption name="send_contact_status_on_update_registration" default="yes">
- <synopsis>Enable sending AMI ContactStatus event when a device refreshes its registration.</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="SrvLookups">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='srv_lookups']/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="MatchHeader">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match_header']/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="TrustConnectedLine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_connected_line']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendConnectedLine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_connected_line']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ConnectedLineMethod">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='connected_line_method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DirectMediaGlareMitigation">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_glare_mitigation']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DisableDirectMediaOnNat">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disable_direct_media_on_nat']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Callerid">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CalleridPrivacy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_privacy']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CalleridTag">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_tag']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TrustIdInbound">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_inbound']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TrustIdOutbound">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_outbound']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendPai">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_pai']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendRpid">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_rpid']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendDiversion">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_diversion']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Mailboxes">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mailboxes']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AggregateMwi">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aggregate_mwi']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaEncryption">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaEncryptionOptimistic">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption_optimistic']/synopsis/node())"/></para>
- </parameter>
- <parameter name="UseAvpf">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_avpf']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ForceAvp">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_avp']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaUseReceivedTransport">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_use_received_transport']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OneTouchRecording">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='one_touch_recording']/synopsis/node())"/></para>
- </parameter>
- <parameter name="InbandProgress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='inband_progress']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CallGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='call_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="PickupGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='pickup_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NamedCallGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_call_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NamedPickupGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_pickup_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DeviceStateBusyAt">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='device_state_busy_at']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38Udptl">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlEc">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ec']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlMaxdatagram">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_maxdatagram']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FaxDetect">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='fax_detect']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlNat">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_nat']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlIpv6">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ipv6']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ToneZone">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tone_zone']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Language">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='language']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RecordOnFeature">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_on_feature']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RecordOffFeature">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_off_feature']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AllowTransfer">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/></para>
- </parameter>
- <parameter name="UserEqPhone">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SdpOwner">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SdpSession">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_session']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TosAudio">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_audio']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TosVideo">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_video']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CosAudio">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_audio']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CosVideo">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_video']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AllowSubscribe">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_subscribe']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SubMinExpiry">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sub_min_expiry']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FromUser">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_user']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FromDomain">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_domain']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MwiFromUser">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mwi_from_user']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RtpEngine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_engine']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsVerify">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_verify']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsRekey">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_rekey']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCertFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cert_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsPrivateKey">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_private_key']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCipher">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cipher']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCaFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCaPath">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_path']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsSetup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_setup']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SrtpTag32">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='srtp_tag_32']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RedirectMethod">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='redirect_method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SetVar">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='set_var']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MessageContext">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='message_context']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Accountcode">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DeviceState">
- <para>The aggregate device state for this endpoint.</para>
- </parameter>
- <parameter name="ActiveChannels">
- <para>The number of active channels associated with this endpoint.</para>
- </parameter>
- <parameter name="SubscribeContext">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='subscribe_context']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Allowoverlap">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_overlap']/synopsis/node())"/></para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="ContactStatusDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about a contact's status.</synopsis>
- <syntax>
- <parameter name="AOR">
- <para>The AoR that owns this contact.</para>
- </parameter>
- <parameter name="URI">
- <para>This contact's URI.</para>
- </parameter>
- <parameter name="Status">
- <para>This contact's status.</para>
- <enumlist>
- <enum name="Reachable"/>
- <enum name="Unreachable"/>
- </enumlist>
- </parameter>
- <parameter name="RoundtripUsec">
- <para>The round trip time in microseconds.</para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- <parameter name="UserAgent">
- <para>Content of the User-Agent header in REGISTER request</para>
- </parameter>
- <parameter name="RegExpire">
- <para>Absolute time that this contact is no longer valid after</para>
- </parameter>
- <parameter name="ViaAddress">
- <para>IP address:port of the last Via header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="CallID">
- <para>Content of the Call-ID header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="ID">
- <para>The sorcery ID of the contact.</para>
- </parameter>
- <parameter name="AuthenticateQualify">
- <para>A boolean indicating whether a qualify should be authenticated.</para>
- </parameter>
- <parameter name="OutboundProxy">
- <para>The contact's outbound proxy.</para>
- </parameter>
- <parameter name="Path">
- <para>The Path header received on the REGISTER.</para>
- </parameter>
- <parameter name="QualifyFrequency">
- <para>The interval in seconds at which the contact will be qualified.</para>
- </parameter>
- <parameter name="QualifyTimeout">
- <para>The elapsed time in decimal seconds after which an OPTIONS
- message is sent before the contact is considered unavailable.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="EndpointList">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about a contact's status.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'endpoint'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Transport">
- <para>The transport configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="Aor">
- <para>The aor configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="Auths">
- <para>The inbound authentication configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="OutboundAuths">
- <para>The outbound authentication configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="DeviceState">
- <para>The aggregate device state for this endpoint.</para>
- </parameter>
- <parameter name="ActiveChannels">
- <para>The number of active channels associated with this endpoint.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <manager name="PJSIPShowEndpoints" language="en_US">
- <synopsis>
- Lists PJSIP endpoints.
- </synopsis>
- <syntax />
- <description>
- <para>
- Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
- is raised that contains relevant attributes and status information. Once all
- endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointList'])" />
- </list-elements>
- <managerEvent language="en_US" name="EndpointListComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about an endpoint list.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- <manager name="PJSIPShowEndpoint" language="en_US">
- <synopsis>
- Detail listing of an endpoint and its objects.
- </synopsis>
- <syntax>
- <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Endpoint" required="true">
- <para>The endpoint to list.</para>
- </parameter>
- </syntax>
- <description>
- <para>
- Provides a detailed listing of options for a given endpoint. Events are issued
- showing the configuration and status of the endpoint and associated objects. These
- events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
- <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
- <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
- associated (for instance AoRs). Once all detail events have been raised a final
- <literal>EndpointDetailComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='IdentifyDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactStatusDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='TransportDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AorDetail'])" />
- </list-elements>
- <managerEvent language="en_US" name="EndpointDetailComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about endpoint details.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- ***/
- #define MOD_DATA_CONTACT "contact"
- /*! Number of serializers in pool if one not supplied. */
- #define SERIALIZER_POOL_SIZE 8
- /*! Pool of serializers to use if not supplied. */
- static struct ast_taskprocessor *serializer_pool[SERIALIZER_POOL_SIZE];
- static pjsip_endpoint *ast_pjsip_endpoint;
- static struct ast_threadpool *sip_threadpool;
- /*! Local host address for IPv4 */
- static pj_sockaddr host_ip_ipv4;
- /*! Local host address for IPv4 (string form) */
- static char host_ip_ipv4_string[PJ_INET6_ADDRSTRLEN];
- /*! Local host address for IPv6 */
- static pj_sockaddr host_ip_ipv6;
- /*! Local host address for IPv6 (string form) */
- static char host_ip_ipv6_string[PJ_INET6_ADDRSTRLEN];
- 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_wait_servant(NULL, register_service_noref, &module);
- }
- int ast_sip_register_service(pjsip_module *module)
- {
- return ast_sip_push_task_wait_servant(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_wait_servant(NULL, unregister_service_noref, &module);
- }
- void ast_sip_unregister_service(pjsip_module *module)
- {
- ast_sip_push_task_wait_servant(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 AST_SIP_AUTHENTICATION_SUCCESS;
- }
- 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);
- }
- int ast_sip_create_request_with_auth_from_old(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
- pjsip_tx_data *old_request, 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_from_old(auths, challenge, old_request, 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, "Unable 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)
- && strlen(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 int do_cli_dump_endpt(void *v_a)
- {
- struct ast_cli_args *a = v_a;
- ast_pjproject_log_intercept_begin(a->fd);
- pjsip_endpt_dump(ast_sip_get_pjsip_endpoint(), a->argc == 4 ? PJ_TRUE : PJ_FALSE);
- ast_pjproject_log_intercept_end();
- return 0;
- }
- static char *cli_dump_endpt(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
- {
- switch (cmd) {
- case CLI_INIT:
- #ifdef AST_DEVMODE
- e->command = "pjsip dump endpt [details]";
- e->usage =
- "Usage: pjsip dump endpt [details]\n"
- " Dump the res_pjsip endpt internals.\n"
- "\n"
- "Warning: PJPROJECT documents that the function used by this\n"
- "CLI command may cause a crash when asking for details because\n"
- "it tries to access all active memory pools.\n";
- #else
- /*
- * In non-developer mode we will not document or make easily accessible
- * the details option even though it is still available. The user has
- * to know it exists to use it. Presumably they would also be aware of
- * the potential crash warning.
- */
- e->command = "pjsip dump endpt";
- e->usage =
- "Usage: pjsip dump endpt\n"
- " Dump the res_pjsip endpt internals.\n";
- #endif /* AST_DEVMODE */
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
- if (4 < a->argc
- || (a->argc == 4 && strcasecmp(a->argv[3], "details"))) {
- return CLI_SHOWUSAGE;
- }
- ast_sip_push_task_wait_servant(NULL, do_cli_dump_endpt, a);
- return CLI_SUCCESS;
- }
- 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_dump_endpt, "Dump the res_pjsip endpt internals"),
- 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;
- }
- int ast_sip_will_uri_survive_restart(pjsip_sip_uri *uri, struct ast_sip_endpoint *endpoint,
- pjsip_rx_data *rdata)
- {
- pj_str_t host_name;
- int result = 1;
- /* Determine if the contact cannot survive a restart/boot. */
- if (uri->port == rdata->pkt_info.src_port
- && !pj_strcmp(&uri->host,
- pj_cstr(&host_name, rdata->pkt_info.src_name))
- /* We have already checked if the URI scheme is sip: or sips: */
- && PJSIP_TRANSPORT_IS_RELIABLE(rdata->tp_info.transport)) {
- pj_str_t type_name;
- /* Determine the transport parameter value */
- if (!strcasecmp("WSS", rdata->tp_info.transport->type_name)) {
- /* WSS is special, as it needs to be ws. */
- pj_cstr(&type_name, "ws");
- } else {
- pj_cstr(&type_name, rdata->tp_info.transport->type_name);
- }
- if (!pj_stricmp(&uri->transport_param, &type_name)
- && (endpoint->nat.rewrite_contact
- /* Websockets are always rewritten */
- || !pj_stricmp(&uri->transport_param,
- pj_cstr(&type_name, "ws")))) {
- /*
- * The contact was rewritten to the reliable transport's
- * source address. Disconnecting the transport for any
- * reason invalidates the contact.
- */
- result = 0;
- }
- }
- return result;
- }
- int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
- pjsip_sip_uri *sip_uri, char *buf, size_t buf_len)
- {
- char *host = NULL;
- static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
- pjsip_param *x_transport;
- if (!ast_strlen_zero(endpoint->transport)) {
- ast_copy_string(buf, endpoint->transport, buf_len);
- return 0;
- }
- x_transport = pjsip_param_find(&sip_uri->other_param, &x_name);
- if (!x_transport) {
- return -1;
- }
- /* Only use x_transport if the uri host is an ip (4 or 6) address */
- host = ast_alloca(sip_uri->host.slen + 1);
- ast_copy_pj_str(host, &sip_uri->host, sip_uri->host.slen + 1);
- if (!ast_sockaddr_parse(NULL, host, PARSE_PORT_FORBID)) {
- return -1;
- }
- ast_copy_pj_str(buf, &x_transport->value, buf_len);
- return 0;
- }
- int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
- pjsip_tpselector *selector)
- {
- pjsip_sip_uri *uri;
- pjsip_tpselector sel = { .type = PJSIP_TPSELECTOR_NONE, };
- uri = pjsip_uri_get_uri(dlg->target);
- if (!selector) {
- selector = &sel;
- }
- ast_sip_set_tpselector_from_ep_or_uri(endpoint, uri, selector);
- pjsip_dlg_set_transport(dlg, selector);
- return 0;
- }
- 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;
- int local_port;
- char default_user[PJSIP_MAX_URL_SIZE];
- if (ast_strlen_zero(user)) {
- ast_sip_get_default_from_user(default_user, sizeof(default_user));
- user = default_user;
- }
- /* 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 */
- type = pjsip_transport_get_type_from_name(&sip_uri->transport_param);
- if (PJSIP_URI_SCHEME_IS_SIPS(sip_uri)) {
- if (type == PJSIP_TRANSPORT_UNSPECIFIED
- || !(pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE)) {
- type = PJSIP_TRANSPORT_TLS;
- }
- } else if (!sip_uri->transport_param.slen) {
- type = PJSIP_TRANSPORT_UDP;
- } else 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_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_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;
- }
- int ast_sip_set_tpselector_from_transport(const struct ast_sip_transport *transport, pjsip_tpselector *selector)
- {
- RAII_VAR(struct ast_sip_transport_state *, transport_state, NULL, ao2_cleanup);
- transport_state = ast_sip_get_transport_state(ast_sorcery_object_get_id(transport));
- if (!transport_state) {
- ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport state for '%s'\n",
- ast_sorcery_object_get_id(transport));
- 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;
- }
- int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip_tpselector *selector)
- {
- RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
- if (ast_strlen_zero(transport_name)) {
- return 0;
- }
- transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_name);
- if (!transport) {
- ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport '%s'\n",
- transport_name);
- return -1;
- }
- return ast_sip_set_tpselector_from_transport(transport, selector);
- }
- int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
- pjsip_sip_uri *sip_uri, pjsip_tpselector *selector)
- {
- char transport_name[128];
- if (ast_sip_get_transport_name(endpoint, sip_uri, transport_name, sizeof(transport_name))) {
- return 0;
- }
- return ast_sip_set_tpselector_from_transport_name(transport_name, selector);
- }
- 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;
- static 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;
- }
- sip_uri->user_param = STR_PHONE;
- }
- 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,
- "Endpoint '%s': Could not create dialog to invalid URI '%s'. Is endpoint registered and reachable?\n",
- ast_sorcery_object_get_id(endpoint), uri);
- }
- return NULL;
- }
- /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
- dlg->sess_count++;
- ast_sip_dlg_set_transport(endpoint, dlg, &selector);
- if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
- dlg->sess_count--;
- 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);
- if (!dlg->local.info->uri) {
- ast_log(LOG_ERROR,
- "Could not parse URI '%s' for endpoint '%s'\n",
- dlg->local.info_str.ptr, ast_sorcery_object_get_id(endpoint));
- dlg->sess_count--;
- pjsip_dlg_terminate(dlg);
- return NULL;
- }
- dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
- if (!ast_strlen_zero(endpoint->contact_user)) {
- pjsip_sip_uri *sip_uri;
- sip_uri = pjsip_uri_get_uri(dlg->local.contact->uri);
- pj_strdup2(dlg->pool, &sip_uri->user, endpoint->contact_user);
- }
- /* 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);
- 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))) {
- ast_log(LOG_ERROR, "Could not create dialog to endpoint '%s' as outbound proxy URI '%s' is not valid\n",
- ast_sorcery_object_get_id(endpoint), outbound_proxy);
- 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;
- pjsip_contact_hdr *contact_hdr;
- ast_assert(status != NULL);
- contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
- if (!contact_hdr || ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(contact_hdr->uri),
- &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) : "");
- #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
- *status = pjsip_dlg_create_uas_and_inc_lock(pjsip_ua_instance(), rdata, &contact, &dlg);
- #else
- *status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
- #endif
- 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--;
- #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
- pjsip_dlg_dec_lock(dlg);
- #endif
- return dlg;
- }
- int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
- char *transport_type, const char *local_name, int local_port, const char *contact)
- {
- pj_str_t tmp;
- /*
- * Initialize the error list in case there is a parse error
- * in the given packet.
- */
- pj_list_init(&rdata->msg_info.parse_err);
- 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;
- pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&tmp, src_name), &rdata->pkt_info.src_addr);
- pj_sockaddr_set_port(&rdata->pkt_info.src_addr, src_port);
- pjsip_parse_rdata(packet, strlen(packet), rdata);
- if (!rdata->msg_info.msg || !pj_list_empty(&rdata->msg_info.parse_err)) {
- return -1;
- }
- if (!ast_strlen_zero(contact)) {
- pjsip_contact_hdr *contact_hdr;
- contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
- if (contact_hdr) {
- contact_hdr->uri = pjsip_parse_uri(rdata->tp_info.pool, (char *)contact,
- strlen(contact), PJSIP_PARSE_URI_AS_NAMEADDR);
- }
- }
- 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;
- }
- 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)
- {
- return ast_sip_create_rdata_with_contact(rdata, packet, src_name, src_port, transport_type,
- local_name, local_port, NULL);
- }
- /* 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, };
- pjsip_uri *sip_uri;
- const char *fromuser;
- 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_WARNING, "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);
- }
- 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;
- }
- sip_uri = pjsip_parse_uri(pool, remote_uri.ptr, remote_uri.slen, 0);
- if (!sip_uri || (!PJSIP_URI_SCHEME_IS_SIP(sip_uri) && !PJSIP_URI_SCHEME_IS_SIPS(sip_uri))) {
- ast_log(LOG_ERROR, "Unable to create outbound %.*s request to endpoint %s as URI '%s' is not valid\n",
- (int) pj_strlen(&method->name), pj_strbuf(&method->name),
- endpoint ? ast_sorcery_object_get_id(endpoint) : "<none>",
- pj_strbuf(&remote_uri));
- pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
- return -1;
- }
- ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(sip_uri), &selector);
- fromuser = endpoint ? (!ast_strlen_zero(endpoint->fromuser) ? endpoint->fromuser : ast_sorcery_object_get_id(endpoint)) : NULL;
- if (sip_dialog_create_from(pool, &from, fromuser,
- 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),
- endpoint ? ast_sorcery_object_get_id(endpoint) : "<none>");
- 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),
- endpoint ? ast_sorcery_object_get_id(endpoint) : "<none>");
- pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
- return -1;
- }
- pjsip_tx_data_set_transport(*tdata, &selector);
- if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
- pjsip_contact_hdr *contact_hdr;
- pjsip_sip_uri *contact_uri;
- static const pj_str_t HCONTACT = { "Contact", 7 };
- static const pj_str_t HCONTACTSHORT = { "m", 1 };
- contact_hdr = pjsip_msg_find_hdr_by_names((*tdata)->msg, &HCONTACT, &HCONTACTSHORT, NULL);
- if (contact_hdr) {
- contact_uri = pjsip_uri_get_uri(contact_hdr->uri);
- pj_strdup2(pool, &contact_uri->user, endpoint->contact_user);
- }
- }
- /* 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 as outbound proxy URI '%s' is not valid\n",
- (int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint),
- endpoint->outbound_proxy);
- 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 {
- ast_assert(endpoint != NULL);
- return create_out_of_dialog_request(pmethod, endpoint, uri, contact, tdata);
- }
- }
- AST_RWLIST_HEAD_STATIC(supplements, ast_sip_supplement);
- void internal_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);
- }
- }
- int ast_sip_register_supplement(struct ast_sip_supplement *supplement)
- {
- internal_sip_register_supplement(supplement);
- ast_module_ref(ast_module_info->self);
- return 0;
- }
- int internal_sip_unregister_supplement(struct ast_sip_supplement *supplement)
- {
- struct ast_sip_supplement *iter;
- SCOPED_LOCK(lock, &supplements, AST_RWLIST_WRLOCK, AST_RWLIST_UNLOCK);
- int res = -1;
- AST_RWLIST_TRAVERSE_SAFE_BEGIN(&supplements, iter, next) {
- if (supplement == iter) {
- AST_RWLIST_REMOVE_CURRENT(next);
- res = 0;
- break;
- }
- }
- AST_RWLIST_TRAVERSE_SAFE_END;
- return res;
- }
- void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement)
- {
- if (!internal_sip_unregister_supplement(supplement)) {
- ast_module_unref(ast_module_info->self);
- }
- }
- 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
- #define TIMER_INACTIVE 0
- #define TIMEOUT_TIMER2 5
- /*! \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;
- /*! Non-zero if endpt_send_request_cb() was called. */
- unsigned int send_cb_called;
- /*! Timeout timer. */
- pj_timer_entry *timeout_timer;
- /*! Original timeout. */
- pj_int32_t timeout;
- /*! The transmit data. */
- pjsip_tx_data *tdata;
- };
- /*! \internal This function gets called by pjsip when the transaction ends,
- * even if it timed out. The lock prevents a race condition if both the pjsip
- * transaction timer and our own timer expire simultaneously.
- */
- static void endpt_send_request_cb(void *token, pjsip_event *e)
- {
- struct send_request_wrapper *req_wrapper = token;
- unsigned int cb_called;
- /*
- * Needed because we cannot otherwise tell if this callback was
- * called when pjsip_endpt_send_request() returns error.
- */
- req_wrapper->send_cb_called = 1;
- if (e->body.tsx_state.type == PJSIP_EVENT_TIMER) {
- ast_debug(2, "%p: PJSIP tsx timer expired\n", req_wrapper);
- if (req_wrapper->timeout_timer
- && req_wrapper->timeout_timer->id != TIMEOUT_TIMER2) {
- ast_debug(3, "%p: Timeout already handled\n", req_wrapper);
- ao2_ref(req_wrapper, -1);
- return;
- }
- } else {
- ast_debug(2, "%p: PJSIP tsx response received\n", req_wrapper);
- }
- ao2_lock(req_wrapper);
- /* It's possible that our own timer was already processing while
- * we were waiting on the lock so check the timer id. If it's
- * still TIMER2 then we still need to process.
- */
- if (req_wrapper->timeout_timer
- && req_wrapper->timeout_timer->id == TIMEOUT_TIMER2) {
- int timers_cancelled = 0;
- ast_debug(3, "%p: Cancelling timer\n", req_wrapper);
- timers_cancelled = pj_timer_heap_cancel_if_active(
- pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
- req_wrapper->timeout_timer, TIMER_INACTIVE);
- if (timers_cancelled > 0) {
- /* If the timer was cancelled the callback will never run so
- * clean up its reference to the wrapper.
- */
- ast_debug(3, "%p: Timer cancelled\n", req_wrapper);
- ao2_ref(req_wrapper, -1);
- } else {
- /*
- * If it wasn't cancelled, it MAY be in the callback already
- * waiting on the lock. When we release the lock, it will
- * now know not to proceed.
- */
- ast_debug(3, "%p: Timer already expired\n", req_wrapper);
- }
- }
- cb_called = req_wrapper->cb_called;
- req_wrapper->cb_called = 1;
- ao2_unlock(req_wrapper);
- /* It's possible that our own timer expired and called the callbacks
- * so no need to call them again.
- */
- if (!cb_called && req_wrapper->callback) {
- req_wrapper->callback(req_wrapper->token, e);
- ast_debug(2, "%p: Callbacks executed\n", req_wrapper);
- }
- ao2_ref(req_wrapper, -1);
- }
- /*! \internal This function gets called by our own timer when it expires.
- * If the timer is cancelled however, the function does NOT get called.
- * The lock prevents a race condition if both the pjsip transaction timer
- * and our own timer expire simultaneously.
- */
- static void send_request_timer_callback(pj_timer_heap_t *theap, pj_timer_entry *entry)
- {
- struct send_request_wrapper *req_wrapper = entry->user_data;
- unsigned int cb_called;
- ast_debug(2, "%p: Internal tsx timer expired after %d msec\n",
- req_wrapper, req_wrapper->timeout);
- ao2_lock(req_wrapper);
- /*
- * If the id is not TIMEOUT_TIMER2 then the timer was cancelled
- * before we got the lock or it was already handled so just clean up.
- */
- if (entry->id != TIMEOUT_TIMER2) {
- ao2_unlock(req_wrapper);
- ast_debug(3, "%p: Timeout already handled\n", req_wrapper);
- ao2_ref(req_wrapper, -1);
- return;
- }
- entry->id = TIMER_INACTIVE;
- ast_debug(3, "%p: Timer handled here\n", req_wrapper);
- cb_called = req_wrapper->cb_called;
- req_wrapper->cb_called = 1;
- ao2_unlock(req_wrapper);
- if (!cb_called && req_wrapper->callback) {
- pjsip_event event;
- PJSIP_EVENT_INIT_TX_MSG(event, req_wrapper->tdata);
- event.body.tsx_state.type = PJSIP_EVENT_TIMER;
- req_wrapper->callback(req_wrapper->token, &event);
- ast_debug(2, "%p: Callbacks executed\n", req_wrapper);
- }
- ao2_ref(req_wrapper, -1);
- }
- static void send_request_wrapper_destructor(void *obj)
- {
- struct send_request_wrapper *req_wrapper = obj;
- pjsip_tx_data_dec_ref(req_wrapper->tdata);
- ast_debug(2, "%p: wrapper destroyed\n", req_wrapper);
- }
- 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;
- pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
- if (!cb && token) {
- /* Silly. Without a callback we cannot do anything with token. */
- pjsip_tx_data_dec_ref(tdata);
- return PJ_EINVAL;
- }
- /* Create wrapper to detect if the callback was actually called on an error. */
- req_wrapper = ao2_alloc(sizeof(*req_wrapper), send_request_wrapper_destructor);
- if (!req_wrapper) {
- pjsip_tx_data_dec_ref(tdata);
- return PJ_ENOMEM;
- }
- ast_debug(2, "%p: Wrapper created\n", req_wrapper);
- req_wrapper->token = token;
- req_wrapper->callback = cb;
- req_wrapper->timeout = timeout;
- req_wrapper->timeout_timer = NULL;
- req_wrapper->tdata = tdata;
- /* Add a reference to tdata. The wrapper destructor cleans it up. */
- pjsip_tx_data_add_ref(tdata);
- if (timeout > 0) {
- pj_time_val timeout_timer_val = { timeout / 1000, timeout % 1000 };
- req_wrapper->timeout_timer = PJ_POOL_ALLOC_T(tdata->pool, pj_timer_entry);
- ast_debug(2, "%p: Set timer to %d msec\n", req_wrapper, timeout);
- pj_timer_entry_init(req_wrapper->timeout_timer, TIMEOUT_TIMER2,
- req_wrapper, send_request_timer_callback);
- /* We need to insure that the wrapper and tdata are available if/when the
- * timer callback is executed.
- */
- ao2_ref(req_wrapper, +1);
- ret_val = pj_timer_heap_schedule(pjsip_endpt_get_timer_heap(endpt),
- req_wrapper->timeout_timer, &timeout_timer_val);
- if (ret_val != PJ_SUCCESS) {
- ast_log(LOG_ERROR,
- "Failed to set timer. Not sending %.*s request to endpoint %s.\n",
- (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>");
- ao2_t_ref(req_wrapper, -2, "Drop timer and routine ref");
- pjsip_tx_data_dec_ref(tdata);
- return ret_val;
- }
- }
- ast_sip_record_request_serializer(tdata);
- /* We need to insure that the wrapper and tdata are available when the
- * transaction callback is executed.
- */
- ao2_ref(req_wrapper, +1);
- ret_val = pjsip_endpt_send_request(endpt, tdata, -1, req_wrapper, endpt_send_request_cb);
- if (ret_val != PJ_SUCCESS) {
- char errmsg[PJ_ERR_MSG_SIZE];
- if (!req_wrapper->send_cb_called) {
- /* endpt_send_request_cb is not expected to ever be called now. */
- ao2_ref(req_wrapper, -1);
- }
- /* 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>");
- if (timeout > 0) {
- int timers_cancelled;
- ao2_lock(req_wrapper);
- timers_cancelled = pj_timer_heap_cancel_if_active(
- pjsip_endpt_get_timer_heap(endpt),
- req_wrapper->timeout_timer, TIMER_INACTIVE);
- if (timers_cancelled > 0) {
- ao2_ref(req_wrapper, -1);
- }
- /* 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 so we claim it is called so our caller can free
- * any resources associated with token because of
- * failure.
- */
- req_wrapper->cb_called = 1;
- }
- ao2_unlock(req_wrapper);
- } else if (req_wrapper->cb_called) {
- /*
- * We cannot report any error. The callback has
- * already freed any resources associated with
- * token.
- */
- ret_val = PJ_SUCCESS;
- }
- }
- 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);
- }
- int ast_sip_send_out_of_dialog_request(pjsip_tx_data *tdata,
- struct ast_sip_endpoint *endpoint, int timeout, 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, timeout, 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 ast_sip_send_out_of_dialog_request(tdata, endpoint, -1, 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_group_named(const char *name, struct ast_serializer_shutdown_group *shutdown_group)
- {
- return ast_threadpool_serializer_group(name, sip_threadpool, shutdown_group);
- }
- struct ast_taskprocessor *ast_sip_create_serializer_group(struct ast_serializer_shutdown_group *shutdown_group)
- {
- char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
- /* Create name with seq number appended. */
- ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip-group-serializer");
- return ast_sip_create_serializer_group_named(tps_name, shutdown_group);
- }
- struct ast_taskprocessor *ast_sip_create_serializer_named(const char *name)
- {
- return ast_sip_create_serializer_group_named(name, NULL);
- }
- struct ast_taskprocessor *ast_sip_create_serializer(void)
- {
- char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
- /* Create name with seq number appended. */
- ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip-serializer");
- return ast_sip_create_serializer_group_named(tps_name, NULL);
- }
- /*!
- * \internal
- * \brief Shutdown the serializers in the default pool.
- * \since 14.0.0
- *
- * \return Nothing
- */
- static void serializer_pool_shutdown(void)
- {
- int idx;
- for (idx = 0; idx < SERIALIZER_POOL_SIZE; ++idx) {
- ast_taskprocessor_unreference(serializer_pool[idx]);
- serializer_pool[idx] = NULL;
- }
- }
- /*!
- * \internal
- * \brief Setup the serializers in the default pool.
- * \since 14.0.0
- *
- * \retval 0 on success.
- * \retval -1 on error.
- */
- static int serializer_pool_setup(void)
- {
- char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
- int idx;
- for (idx = 0; idx < SERIALIZER_POOL_SIZE; ++idx) {
- /* Create name with seq number appended. */
- ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip/default");
- serializer_pool[idx] = ast_sip_create_serializer_named(tps_name);
- if (!serializer_pool[idx]) {
- serializer_pool_shutdown();
- return -1;
- }
- }
- return 0;
- }
- static struct ast_taskprocessor *serializer_pool_pick(void)
- {
- int idx;
- int pos = 0;
- if (!serializer_pool[0]) {
- return NULL;
- }
- for (idx = 1; idx < SERIALIZER_POOL_SIZE; ++idx) {
- if (ast_taskprocessor_size(serializer_pool[idx]) < ast_taskprocessor_size(serializer_pool[pos])) {
- pos = idx;
- }
- }
- return serializer_pool[pos];
- }
- int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
- {
- if (!serializer) {
- serializer = serializer_pool_pick();
- }
- return ast_taskprocessor_push(serializer, 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_wait()
- * 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;
- }
- static int ast_sip_push_task_wait(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
- {
- /* This method is an onion */
- struct sync_task_data std;
- 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 (ast_sip_push_task(serializer, 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;
- }
- int ast_sip_push_task_wait_servant(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
- {
- if (ast_sip_thread_is_servant()) {
- return sip_task(task_data);
- }
- return ast_sip_push_task_wait(serializer, sip_task, task_data);
- }
- int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
- {
- return ast_sip_push_task_wait_servant(serializer, sip_task, task_data);
- }
- int ast_sip_push_task_wait_serializer(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data)
- {
- if (!serializer) {
- /* Caller doesn't care which PJSIP serializer the task executes under. */
- serializer = serializer_pool_pick();
- if (!serializer) {
- /* No serializer picked to execute the task */
- return -1;
- }
- }
- if (ast_taskprocessor_is_task(serializer)) {
- /*
- * We are the requested serializer so we must execute
- * the task now or deadlock waiting on ourself to
- * execute it.
- */
- return sip_task(task_data);
- }
- return ast_sip_push_task_wait(serializer, sip_task, task_data);
- }
- 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)
- {
- pj_status_t status;
- supplement_outgoing_response(tdata, sip_endpoint);
- status = pjsip_endpt_send_response(ast_sip_get_pjsip_endpoint(), res_addr, tdata, NULL, NULL);
- if (status != PJ_SUCCESS) {
- pjsip_tx_data_dec_ref(tdata);
- }
- return status == PJ_SUCCESS ? 0 : -1;
- }
- 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) {
- struct ast_sip_contact *contact;
- /* ast_sip_create_response bumps the refcount of the contact and adds it to the tdata.
- * We'll leak that reference if we don't get rid of it here.
- */
- contact = ast_sip_mod_data_get(tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT);
- ao2_cleanup(contact);
- ast_sip_mod_data_set(tdata->pool, tdata->mod_data, supplement_module.id, MOD_DATA_CONTACT, NULL);
- 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;
- }
- int ast_sip_get_host_ip(int af, pj_sockaddr *addr)
- {
- if (af == pj_AF_INET() && !ast_strlen_zero(host_ip_ipv4_string)) {
- pj_sockaddr_copy_addr(addr, &host_ip_ipv4);
- return 0;
- } else if (af == pj_AF_INET6() && !ast_strlen_zero(host_ip_ipv6_string)) {
- pj_sockaddr_copy_addr(addr, &host_ip_ipv6);
- return 0;
- }
- return -1;
- }
- const char *ast_sip_get_host_ip_string(int af)
- {
- if (af == pj_AF_INET()) {
- return host_ip_ipv4_string;
- } else if (af == pj_AF_INET6()) {
- return host_ip_ipv6_string;
- }
- return NULL;
- }
- int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
- char *buf, size_t buf_len)
- {
- switch (dtmf) {
- case AST_SIP_DTMF_NONE:
- ast_copy_string(buf, "none", buf_len);
- break;
- case AST_SIP_DTMF_RFC_4733:
- ast_copy_string(buf, "rfc4733", buf_len);
- break;
- case AST_SIP_DTMF_INBAND:
- ast_copy_string(buf, "inband", buf_len);
- break;
- case AST_SIP_DTMF_INFO:
- ast_copy_string(buf, "info", buf_len);
- break;
- case AST_SIP_DTMF_AUTO:
- ast_copy_string(buf, "auto", buf_len);
- break;
- case AST_SIP_DTMF_AUTO_INFO:
- ast_copy_string(buf, "auto_info", buf_len);
- break;
- default:
- buf[0] = '\0';
- return -1;
- }
- return 0;
- }
- int ast_sip_str_to_dtmf(const char * dtmf_mode)
- {
- int result = -1;
- if (!strcasecmp(dtmf_mode, "info")) {
- result = AST_SIP_DTMF_INFO;
- } else if (!strcasecmp(dtmf_mode, "rfc4733")) {
- result = AST_SIP_DTMF_RFC_4733;
- } else if (!strcasecmp(dtmf_mode, "inband")) {
- result = AST_SIP_DTMF_INBAND;
- } else if (!strcasecmp(dtmf_mode, "none")) {
- result = AST_SIP_DTMF_NONE;
- } else if (!strcasecmp(dtmf_mode, "auto")) {
- result = AST_SIP_DTMF_AUTO;
- } else if (!strcasecmp(dtmf_mode, "auto_info")) {
- result = AST_SIP_DTMF_AUTO_INFO;
- }
- return result;
- }
- /*!
- * \brief Set name and number information on an identity header.
- *
- * \param pool Memory pool to use for string duplication
- * \param id_hdr A From, P-Asserted-Identity, or Remote-Party-ID header to modify
- * \param id The identity information to apply to the header
- */
- void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr, const struct ast_party_id *id)
- {
- pjsip_name_addr *id_name_addr;
- pjsip_sip_uri *id_uri;
- id_name_addr = (pjsip_name_addr *) id_hdr->uri;
- id_uri = pjsip_uri_get_uri(id_name_addr->uri);
- if (id->name.valid) {
- if (!ast_strlen_zero(id->name.str)) {
- int name_buf_len = strlen(id->name.str) * 2 + 1;
- char *name_buf = ast_alloca(name_buf_len);
- ast_escape_quoted(id->name.str, name_buf, name_buf_len);
- pj_strdup2(pool, &id_name_addr->display, name_buf);
- } else {
- pj_strdup2(pool, &id_name_addr->display, NULL);
- }
- }
- if (id->number.valid) {
- pj_strdup2(pool, &id_uri->user, id->number.str);
- }
- }
- 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);
- }
- }
- long ast_sip_threadpool_queue_size(void)
- {
- return ast_threadpool_queue_size(sip_threadpool);
- }
- #ifdef TEST_FRAMEWORK
- AST_TEST_DEFINE(xml_sanitization_end_null)
- {
- char sanitized[8];
- switch (cmd) {
- case TEST_INIT:
- info->name = "xml_sanitization_end_null";
- info->category = "/res/res_pjsip/";
- info->summary = "Ensure XML sanitization works as expected with a long string";
- info->description = "This test sanitizes a string which exceeds the output\n"
- "buffer size. Once done the string is confirmed to be NULL terminated.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
- ast_sip_sanitize_xml("aaaaaaaaaaaa", sanitized, sizeof(sanitized));
- if (sanitized[7] != '\0') {
- ast_test_status_update(test, "Sanitized XML string is not null-terminated when it should be\n");
- return AST_TEST_FAIL;
- }
- return AST_TEST_PASS;
- }
- AST_TEST_DEFINE(xml_sanitization_exceeds_buffer)
- {
- char sanitized[8];
- switch (cmd) {
- case TEST_INIT:
- info->name = "xml_sanitization_exceeds_buffer";
- info->category = "/res/res_pjsip/";
- info->summary = "Ensure XML sanitization does not exceed buffer when output won't fit";
- info->description = "This test sanitizes a string which before sanitization would\n"
- "fit within the output buffer. After sanitization, however, the string would\n"
- "exceed the buffer. Once done the string is confirmed to be NULL terminated.";
- return AST_TEST_NOT_RUN;
- case TEST_EXECUTE:
- break;
- }
- ast_sip_sanitize_xml("<><><>&", sanitized, sizeof(sanitized));
- if (sanitized[7] != '\0') {
- ast_test_status_update(test, "Sanitized XML string is not null-terminated when it should be\n");
- return AST_TEST_FAIL;
- }
- return AST_TEST_PASS;
- }
- #endif
- /*!
- * \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;
- }
- void internal_res_pjsip_ref(void)
- {
- ast_module_ref(ast_module_info->self);
- }
- void internal_res_pjsip_unref(void)
- {
- ast_module_unref(ast_module_info->self);
- }
- static int unload_pjsip(void *data)
- {
- /*
- * These calls need the pjsip endpoint and serializer to clean up.
- * If they're not set, then there's nothing to clean up anyway.
- */
- if (ast_pjsip_endpoint && serializer_pool[0]) {
- ast_res_pjsip_cleanup_options_handling();
- internal_sip_destroy_outbound_authentication();
- ast_res_pjsip_cleanup_message_filter();
- ast_sip_destroy_distributor();
- ast_sip_destroy_transport_management();
- ast_res_pjsip_destroy_configuration();
- ast_sip_destroy_system();
- ast_sip_destroy_global_headers();
- internal_sip_unregister_service(&supplement_module);
- ast_sip_destroy_transport_events();
- }
- if (monitor_thread) {
- stop_monitor_thread();
- monitor_thread = NULL;
- }
- if (memory_pool) {
- /* This mimics the behavior of pj_pool_safe_release
- * which was introduced in pjproject 2.6.
- */
- pj_pool_t *temp_pool = memory_pool;
- memory_pool = NULL;
- pj_pool_release(temp_pool);
- }
- ast_pjsip_endpoint = NULL;
- if (caching_pool.lock) {
- ast_pjproject_caching_pool_destroy(&caching_pool);
- }
- pj_shutdown();
- return 0;
- }
- static int load_pjsip(void)
- {
- const unsigned int flags = 0; /* no port, no brackets */
- pj_status_t status;
- /* The third parameter is just copied from
- * example code from PJLIB. This can be adjusted
- * if necessary.
- */
- ast_pjproject_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");
- goto error;
- }
- /* 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");
- goto error;
- }
- if (!pj_gethostip(pj_AF_INET(), &host_ip_ipv4)) {
- pj_sockaddr_print(&host_ip_ipv4, host_ip_ipv4_string, sizeof(host_ip_ipv4_string), flags);
- ast_verb(3, "Local IPv4 address determined to be: %s\n", host_ip_ipv4_string);
- }
- if (!pj_gethostip(pj_AF_INET6(), &host_ip_ipv6)) {
- pj_sockaddr_print(&host_ip_ipv6, host_ip_ipv6_string, sizeof(host_ip_ipv6_string), flags);
- ast_verb(3, "Local IPv6 address determined to be: %s\n", host_ip_ipv6_string);
- }
- 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");
- goto error;
- }
- return AST_MODULE_LOAD_SUCCESS;
- error:
- return AST_MODULE_LOAD_DECLINE;
- }
- /*
- * This is a place holder function to ensure that pjmedia_strerr() is at
- * least directly referenced by this module to ensure that the loader
- * linker will link to the function. If a module only indirectly
- * references a function from another module, such as a callback parameter
- * to a function, the loader linker has been known to miss the link.
- */
- void never_called_res_pjsip(void);
- void never_called_res_pjsip(void)
- {
- pjmedia_strerror(0, NULL, 0);
- }
- static int load_module(void)
- {
- struct ast_threadpool_options options;
- CHECK_PJPROJECT_MODULE_LOADED();
- /* pjproject and config_system need to be initialized before all else */
- if (pj_init() != PJ_SUCCESS) {
- return AST_MODULE_LOAD_DECLINE;
- }
- if (pjlib_util_init() != PJ_SUCCESS) {
- goto error;
- }
- /* Register PJMEDIA error codes for SDP parsing errors */
- if (pj_register_strerror(PJMEDIA_ERRNO_START, PJ_ERRNO_SPACE_SIZE, pjmedia_strerror)
- != PJ_SUCCESS) {
- ast_log(LOG_WARNING, "Failed to register pjmedia error codes. Codes will not be decoded.\n");
- }
- if (ast_sip_initialize_system()) {
- ast_log(LOG_ERROR, "Failed to initialize SIP 'system' configuration section. Aborting load\n");
- goto error;
- }
- /* The serializer needs threadpool and threadpool needs pjproject to be initialized so it's next */
- sip_get_threadpool_options(&options);
- options.thread_start = sip_thread_start;
- sip_threadpool = ast_threadpool_create("SIP", NULL, &options);
- if (!sip_threadpool) {
- goto error;
- }
- if (serializer_pool_setup()) {
- ast_log(LOG_ERROR, "Failed to create SIP serializer pool. Aborting load\n");
- goto error;
- }
- if (ast_sip_initialize_scheduler()) {
- ast_log(LOG_ERROR, "Failed to start scheduler. Aborting load\n");
- goto error;
- }
- /* Now load all the pjproject infrastructure. */
- if (load_pjsip()) {
- goto error;
- }
- if (ast_sip_initialize_transport_events()) {
- ast_log(LOG_ERROR, "Failed to initialize SIP transport monitor. Aborting load\n");
- goto error;
- }
- ast_sip_initialize_dns();
- ast_sip_initialize_global_headers();
- if (ast_res_pjsip_preinit_options_handling()) {
- ast_log(LOG_ERROR, "Failed to pre-initialize OPTIONS handling. Aborting load\n");
- goto error;
- }
- if (ast_res_pjsip_initialize_configuration(ast_module_info)) {
- ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
- goto error;
- }
- if (ast_sip_initialize_transport_management()) {
- ast_log(LOG_ERROR, "Failed to initialize SIP transport management. Aborting load\n");
- goto error;
- }
- if (ast_sip_initialize_distributor()) {
- ast_log(LOG_ERROR, "Failed to register distributor module. Aborting load\n");
- goto error;
- }
- if (internal_sip_register_service(&supplement_module)) {
- ast_log(LOG_ERROR, "Failed to initialize supplement hooks. Aborting load\n");
- goto error;
- }
- if (internal_sip_initialize_outbound_authentication()) {
- ast_log(LOG_ERROR, "Failed to initialize outbound authentication. Aborting load\n");
- goto error;
- }
- if (ast_res_pjsip_init_options_handling(0)) {
- ast_log(LOG_ERROR, "Failed to initialize OPTIONS handling. Aborting load\n");
- goto error;
- }
- if (ast_res_pjsip_init_message_filter()) {
- ast_log(LOG_ERROR, "Failed to initialize message IP updating. Aborting load\n");
- goto error;
- }
- ast_cli_register_multiple(cli_commands, ARRAY_LEN(cli_commands));
- AST_TEST_REGISTER(xml_sanitization_end_null);
- AST_TEST_REGISTER(xml_sanitization_exceeds_buffer);
- ast_pjproject_ref();
- return AST_MODULE_LOAD_SUCCESS;
- error:
- unload_pjsip(NULL);
- /* These functions all check for NULLs and are safe to call at any time */
- ast_sip_destroy_scheduler();
- serializer_pool_shutdown();
- ast_threadpool_shutdown(sip_threadpool);
- return AST_MODULE_LOAD_DECLINE;
- }
- 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_wait_servant(NULL, reload_configuration_task, NULL)) {
- ast_log(LOG_WARNING, "Failed to reload PJSIP\n");
- return -1;
- }
- return 0;
- }
- static int unload_module(void)
- {
- AST_TEST_UNREGISTER(xml_sanitization_end_null);
- AST_TEST_UNREGISTER(xml_sanitization_exceeds_buffer);
- ast_cli_unregister_multiple(cli_commands, ARRAY_LEN(cli_commands));
- /* 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_wait_servant(NULL, unload_pjsip, NULL);
- ast_sip_destroy_scheduler();
- serializer_pool_shutdown();
- ast_threadpool_shutdown(sip_threadpool);
- ast_pjproject_unref();
- 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,
- );
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