1901 Commits (09932a13efa6719b481aeab0402893de230c1947)

Author	SHA1	Message	Date
George Joseph	cf8d1d8cc7	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	9 years ago
George Joseph	35cc916e63	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	9 years ago
Corey Farrell	9c20596400	AST-2017-006: Fix app_minivm application MinivmNotify command injection ... An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	9 years ago
Andrey Egorov	dc4435f68d	res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif ... Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db	9 years ago
Joshua Colp	5eececf786	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO." into 14	9 years ago
Corey Farrell	b9e497d7fe	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. ... This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	9 years ago
Sean Bright	c663d59acd	basic-pbx: Remove res_pjsip_multihomed from sample config ... ASTERISK-27127 #close Reported by: HZMI8gkCvPpom0tM Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789	9 years ago
George Joseph	9bd8812337	res_musiconhold: Add kill_escalation_delay, kill_method to class ... By default, when res_musiconhold reloads or unloads, it sends a HUP signal to custom applications (and all descendants), waits 100ms, then sends a TERM signal, waits 100ms, then finally sends a KILL signal. An application which is interacting with an external device and/or spawns children of its own may not be able to exit cleanly in the default times, expecially if sent a KILL signal, or if it's children are getting signals directly from res_musiconhoild. * To allow extra time, the 'kill_escalation_delay' class option can be used to set the number of milliseconds res_musiconhold waits before escalating kill signals, with the default being the current 100ms. * To control to whom the signals are sent, the "kill_method" class option can be set to "process_group" (the default, existing behavior), which sends signals to the application and its descendants directly, or "process" which sends signals only to the application itself. Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b	9 years ago
George Joseph	be0feb2ce8	chan_pjsip: Fix ability to send UPDATE on COLP ... When connected_line_method is "invite", we're supposed to determine if the client can support UPDATE and if it can, send UPDATE instead of INVITE to avoid the SDP renegotiation. Not only was pjproject not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing that invite_tsx wasn't NULL which isn't always the case. * Updated chan_pjsip/update_connected_line_information to drop the requirement that invite_tsx isn't NULL. * Submitted patch to pjproject sip_inv.c that sets the PJSIP_INV_SUPPORT_UPDATE flag correctly. * Updated pjsip.conf.sample to clarify what happens when "invite" is specified. ASTERISK-27095 Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560	9 years ago
Alexei Gradinari	227f3fe93f	app_voicemail: IMAP connection control ... A new global option "imap_poll_logout" was added to specify whether need to disconnect from the IMAP server after polling of mailboxes. ASTERISK-27068 #close Closing IMAP connection after loading mailbox from voicemail.conf ASTERISK-24052 #close Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a	9 years ago
Rodrigo Ramírez Norambuena	8fac4b77db	cdr: fix mistake spelling of a word for Unanswered. ... Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	9 years ago
Alexei Gradinari	3f61d56ddf	res_pjsip: New endpoint option "notify_early_inuse_ringing" ... This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	9 years ago
Sean Bright	f7db391740	codecs.conf.sample: Fix max_bandwidth speling error ... Reported by Sylvain Boily via asterisk-dev mailing list. Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d	9 years ago
Jenkins2	665cfd99dc	Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 14	9 years ago
Rodrigo Ramírez Norambuena	fca75547e5	Fix spelling queues.conf.sample file ... Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee	9 years ago
Joshua Colp	85127e8f8d	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 14	9 years ago
Alexei Gradinari	d02020791b	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	9 years ago
Joshua Colp	097f90220a	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	9 years ago
George Joseph	944f9435bc	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Richard Mudgett	f592947b98	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	9 years ago
Walter Doekes	09252e9d59	samples: Undo removal of include from canonicalize-app-names commit. ... This include was accidentally removed in changeset Ia79aea64de89531362e993e34230c2044a70aa93. My bad. Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082	9 years ago
Joshua Colp	2bedb0f996	Merge "pjsip: Add Alembic for PUBLISH support." into 14	9 years ago
Joshua Colp	7942378c55	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	9 years ago
Walter Doekes	c5533488cc	samples: Canonicalize app names in extensions.conf.sample. ... This takes care of warnings by ossobv/asterisklint. Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93	9 years ago
George Joseph	baf7660c7b	sample_config: Add samples for pubsub to pjsip.conf.sample ... Added: * outbound-publish * resource_list * inbound-publication * asterisk-publication Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a	9 years ago
George Joseph	27344675be	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff7)	9 years ago
Sean Bright	63a79e395a	res_musiconhold: Document the 'format' option ... ASTERISK-26086 #close Reported by: Jens Bürger Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e	9 years ago
Joshua Colp	b872a73e6e	Merge "rtp_engine: allocate RTP dynamic payloads per session" into 14	9 years ago
Kevin Harwell	d0649254b5	rtp_engine: allocate RTP dynamic payloads per session ... Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "no" in Asterisk 14. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000) Change-Id: If282877199b85da8dde7eb9452cdedaa19da586a	9 years ago
Richard Begg	d7f50afb7c	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	9 years ago
Sean Bright	217ba2e5d0	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	9 years ago
George Joseph	6541a07083	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. 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. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	9 years ago
Matt Jordan	d31aa48d09	configs/samples/hep.conf.sample: Clarify how the HEP stack works ... This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124	9 years ago
Daniel Journo	893628abdd	app_voicemail: Cannot set fromstring on a per-mailbox basis ... * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	9 years ago
Joshua Colp	af12598253	Merge "config: Improve documentation and behavior of outbound_proxy option." into 14	9 years ago
Tzafrir Cohen	c9f9eece87	pjsip.conf.sample: user_agent: not a specific version ... Use the description of useragent from sip.conf here. ASTERISK-26825 #close Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755	9 years ago
Joshua Colp	9c01a04261	config: Improve documentation and behavior of outbound_proxy option. ... This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593	9 years ago
Richard Mudgett	8a6cf4818f	res_pjsip: Update authentication realm documentation. ... 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. An empty inbound auth realm represents the global section's default_realm value when the authentication object is used to challenge an incoming request. An empty outgoing auth realm is treated as a don't care wildcard when the authentication object is used to respond to an incoming authentication challenge. ASTERISK-26799 Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce	9 years ago
Sean Bright	23ee13dcd1	app_voicemail: Allow 'Comedian Mail' branding to be overriden ... Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6	9 years ago
George Joseph	abf9561518	configs/samples: Fix placement of 'identify' entry in sorcery.conf ... The entry for 'identify' was incorrectly placed in the res_pjsip section when it should be in res_pjsip_endpoint_identifier_ip. ASTERISK-26785 #close Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a	9 years ago
Sean Bright	bd0258b0b1	res_odbc: Remove deprecated settings from sample configuration file ... ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f	9 years ago
George Joseph	84aadd4e81	debug_utilities: Create ast_loggrabber ... ast_loggrabber gathers log files from customizable search patterns, optionally converts POSIX timestamps to a readable format and tarballs the results. Also a few tweaks were made to ast_coredumper. Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 (cherry picked from commit c709152878)	9 years ago
George Joseph	676865c2fd	debug_utilities: Create the ast_coredumper utility ... This utility allows easy manipulation of asterisk coredumps. * Configurable search paths and patterns for existing coredumps * Can generate a consistent coredump from the running instance * Can dump the lock_infos table from a coredump * Dumps backtraces to separate files... - thread apply 1 bt full -> <coredump>.thread1.txt - thread apply all bt -> <coredump>.brief.txt - thread apply all bt full -> <coredump>.full.txt - lock_infos table -> <coredump>.locks.txt * Can tarball corefiles and optionally delete them after processing * Can tarball results files and optionally delete them after processing * Converts ':' in coredump and results file names '-' to facilitate uploading. Jira for instance, won't accept file names with colons in them. Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. [1] For *BSDs, the "devel/gdb" package might have to be installed to get a recent gdb. The utility will check all instances of gdb it finds in $PATH and if one isn't found that can run python, it prints a friendly error. Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd (cherry picked from commit cb47b45560)	9 years ago
Richard Mudgett	846c77b28c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Sebastien Duthil	eb5077fb26	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	9 years ago
Alexander Traud	6445f21caa	rtp_engine: Allow more than 32 dynamic payload types. ... Since adding all remaining rates of Signed Linear (ASTERISK-24274) and SILK (Gerrit 3136), only one RTP Payload Type is left in the dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964	9 years ago
Kevin Harwell	35f9d472ba	codecs.conf.sample: Add sample and option descriptions for codec_opus ... codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b	9 years ago
Rusty Newton	c2a2643c69	SAC documentation: don't specify transports for endpoints and registrations ... Removing explicit transport definition for endpoints and registrations. It isn't necessary and isn't generally advised. ASTERISK-26514 #close Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb	9 years ago
Joshua Colp	791d2319ce	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
zuul	7c3cc1b605	Merge "pjsip: Support dual stack automatically." into 14	9 years ago