asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua C. Colp	4e7adbd8f4	res_pjsip_pubsub: Add ability to persist generator state information. Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de	6 years ago
Joshua Colp	e89c8bc0d2	Merge "sig_pri: Fix deadlock caused by sig_pri_queue_hangup"	6 years ago
Joshua Colp	bf0247ae7c	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic"	6 years ago
Sean Bright	312abaa1fe	res_pjsip_endpoint_identifier_ip.c: Add port matching support Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	6 years ago
George Joseph	a4fd89536d	Merge "app_bridgeaddchan.c: Make BridgeAdd be more like Bridge"	6 years ago
George Joseph	6b7334a311	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail."	6 years ago
George Joseph	d66b01d3bf	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings"	6 years ago
George Joseph	ab5b97c0d3	Merge "features.c: Make Bridge application tolerate unspecified channel."	6 years ago
Friendly Automation	5b815fe1ac	Merge "app_dial.c: Simplify dialplan using Dial."	6 years ago
Friendly Automation	5050c45e06	Merge "app_page.c: Simplify dialplan using Page."	6 years ago
Joshua Colp	b2664fd3a4	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message."	6 years ago
Friendly Automation	1aa294f6f5	Merge "app_chanspy.c: Reduce log message level from notice to verbose."	6 years ago
Friendly Automation	255a647c53	Merge "websocket: Consider pending SSL data when waiting for socket input"	6 years ago
George Joseph	cc2d1f2545	Merge "contrib/valgrind: Fix use of frame-level suppression"	6 years ago
George Joseph	ee7d72eb72	sig_pri: Fix deadlock caused by sig_pri_queue_hangup The change to add setting hangupsource to sig_pri_queue_hangup() made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued deadlocks when a hangup request was received from the core at the same time a hanguprequest was received from the remote end via the D channel. Although the PRI's channel private structure was being unlocked before setting the hangupsource, the PRI's own lock was still being held during the process. If channel actions were also coming from the core, a deadlock on the PRI could result. This deadlock could then escalate to the entire DAHDI subsystem via DAHDI's global interface list lock, especially if someone used the PRI CLI commands. Fix: * We now unlock the PRI as well as the PRI's channel private structure before setting the hangupsource, then relock both afterwards. ASTERISK-28605 Reported by: Dirk Wendland Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9	6 years ago
Richard Mudgett	fe3cce816c	app_chanisavail.c: Simplify dialplan using ChanIsAvail. Dialplan has to be careful about passing an empty device list or empty positions in the list. As a result, dialplan has to check for these conditions before using ChanIsAvail. Simplify dialplan by making ChanIsAvail handle these conditions gracefully. * Made tolerate empty positions in the device list. * Simplified the code and eliminated some unnecessary indention. ASTERISK-28638 Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3	6 years ago
George Joseph	1c9ddad4db	stasis.c: Use correct topic name in stasis_topic_pool_delete_topic When a topic is created for an object, its name is only <object>:<uniqueid> For example: bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 When a topic is added to a pool, its name has the pool's topic name prepended. For example: bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 The topic_pool_entry's name however, is only what was passed in to stasis_topic_pool_get_topic which is bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 That's actually correct because the entry is qualified by the pool that's in. When you're ready to delete the entry from the pool, you retrieve the tropic name from the object but since it now has the pool's topic name prepended, it won't be found in the pool container. Fix: * Modified stasis_topic_pool_delete_topic() to skip past the pool topic's name, if it was prepended to the topic name, before searching the container for a pool entry. ASTERISK-28633 Reported by: Joeran Vinzens Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6	6 years ago
Richard Mudgett	19069f7db7	app_bridgeaddchan.c: Make BridgeAdd be more like Bridge * Made BridgeAdd not hangup the call if there is a problem. * Reduced message level from warning to verbose for normal exception cases. * Added a loop safety check to BridgeAdd. * Made BridgeAdd set BRIDGERESULT with the status when dialplan is resumed. Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426	6 years ago
Richard Mudgett	abcb4ab321	app_dial.c: Simplify dialplan using Dial. Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Dial. Simplify dialplan by making Dial handle these conditions gracefully. * Made tolerate empty positions in the dialed device list. * Reduced some message log levels from notice to verbose. ASTERISK-28638 Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9	6 years ago
Richard Mudgett	d86a6ac5ce	app_page.c: Simplify dialplan using Page. Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	6 years ago
Richard Mudgett	0376f2bba9	features.c: Make Bridge application tolerate unspecified channel. The Bridge application was inconsistent if the channel to bridge with is not specified. If no parameters are given then a warning is issued and the current channel is hung up. If options are given but no channel is specified then a warning is issued and the current channel is not hung up. * Made the Bridge application give a verbose message instead of a warning if the channel to bridge with is not specified and made not hang up the current channel. As a result dialplan no longer needs to check if a channel name is passed before calling Bridge and simply needs to check the BRIDGERESULT channel variable instead. This is something you likely want your dialplan to do anyway. * Fixed up L() option warning message. It is up to the caller to determine if the channel is hung up because of the warning. Dial() hangs up the current channel while Bridge() does not. Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73	6 years ago
Richard Mudgett	0d1f3d9bf3	app_chanspy.c: Reduce log message level from notice to verbose. Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	a457947198	app_softhangup.c: Reduce unnecessary warning to verbose message. Why log a warning for something your dialplan explicitly asked for? Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7	6 years ago
Sean Bright	b40dd11afe	res_pjsip_config_wizard: Fix change detection for wizard settings ast_sorcery_changeset_create() is not commutative and will fail to detect differences between two variable lists depending on what changed, so switch to ast_variable_lists_match(). ASTERISK-28492 #close Reported by: Jean-Denis Girard Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1	6 years ago
Sean Bright	7d94bdde9d	res_agi: Improve GET FULL VARIABLE documentation ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	6 years ago
Sean Bright	87110c1bdf	websocket: Consider pending SSL data when waiting for socket input When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
Joshua Colp	cee68ea4f2	Merge "func_odbc: acf_odbc_read() and cli_odbc_read() unicode support"	6 years ago
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	6 years ago
George Joseph	be93537382	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option"	6 years ago
Friendly Automation	2a8f759374	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP"	6 years ago
Sean Bright	fc99ac8c9a	db: Initialize condition primitive before use The db_init() function ultimately calls db_sync() which signals the condition before it is initialized. Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab	6 years ago
George Joseph	7c4fc2e39a	Merge "config.c: Skip UTF-8 BOMs if present when reading config files"	6 years ago
Joshua C. Colp	dd823e50e4	Merge "main/file.c: Limit media cache usage to remote files."	6 years ago
Joshua C. Colp	d21427cadd	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR."	6 years ago
Friendly Automation	c3cf0e330c	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors."	6 years ago
Friendly Automation	07b7c6f50a	Merge "confbridge: Add support for specifying maximum sample rate."	6 years ago
Sean Bright	40b5cf8f52	config.c: Skip UTF-8 BOMs if present when reading config files ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
Kevin Reeves	c626ccec12	main/file.c: Limit media cache usage to remote files. When testing for the existance of a file, the media cache is searched even if the file has no chance of being in it. This can cause performance issues as the media cache size increases. As a result, calls to applications like Read and Playback using local files must scan through the media cache before playing. Under load and with a large cache, this can delay the playback of those files. This patch updates the function that checks for the existance of a file to only consult the media cache database if the requested file is a remote path. It introduces a new is_remote_path() function in main/file.c. ASTERISK-28625 #close Reported-by: kevin@phoneburner.com Change-Id: If91137493732d9034dafa381c081c69274a7dcc9	6 years ago
George Joseph	657ada8bfd	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read"	6 years ago
Friendly Automation	308592a681	Merge "sip_to_pjsip.py: Fix trustrpid typo"	6 years ago
Joshua C. Colp	c29b7c00e5	Merge "configure: Add check for MySQL client bool and my_bool type usage."	6 years ago
Joshua C. Colp	2779189689	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP."	6 years ago
Snuffy	095c204fe0	contrib/valgrind: Fix use of frame-level suppression Fix use of frame-level wildcard usage in suppression file. ASTERISK-27243 #close Reported-by: Richard Kenner Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51	6 years ago
Pascal Cadotte Michaud	e494d5fd76	sip_to_pjsip.py: Fix trustrpid typo ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	6 years ago
Frederic LE FOLL	a83625b366	app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. Temporary channel lifespan is very short and CDR deactivation request through ast_cdr_set_property() may happen when CDR is not available yet. Use CDR_PROP() dialplan function instead, it will first wait for pending CDR insertion requests to be processed. ASTERISK-28636 Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074	6 years ago
George Joseph	8587ef3a0c	Merge "res_pjsip_nat: Restore original contact for REGISTER responses"	6 years ago
Joshua C. Colp	ed394ce5b1	configure: Add check for MySQL client bool and my_bool type usage. Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Joshua C. Colp	89b7144fbd	confbridge: Add support for specifying maximum sample rate. ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Friendly Automation	7e441d6b25	Merge "PJSIP_CONTACT: add missing argument documentation"	6 years ago
Joshua C. Colp	a603d7d324	res_pjsip_session: Set stream state on created streams for incoming SDP. A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago

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