asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	bfc93cc954	chan_pjsip: Ignore RTP that we haven't negotiated If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now keep track of the formats that have been negotiated and check before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3	5 years ago
Joshua Colp	9ba3bddd37	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY" into 13	5 years ago
Joshua Colp	3a135e5827	Merge "app_queue: Deprecate the QueueMemberPause.Reason field" into 13	5 years ago
Sean Bright	1a4fcaadb2	app_queue: Deprecate the QueueMemberPause.Reason field The QueueMemberPause AMI event includes two fields that return the reason a member was paused. * In release branches, deprecate Reason in favor of PausedReason. * In master, remove the Reason field entirely. ASTERISK-28349 #close Reported by: Niksa Baldun Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296	5 years ago
Joshua Colp	9b22061e2c	Merge "res_pjsip_endpoint_identifier_ip: Document support for hostnames" into 13	5 years ago
Joshua Colp	a902641833	Merge "func_curl: Add 'followlocation' option to CURLOPT()" into 13	5 years ago
Joshua Colp	8bf22b8abb	Merge "app_record: Do not hang up if beep audio is missing" into 13	5 years ago
Joshua Colp	0d1fcf0626	Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr." into 13	5 years ago
Sean Bright	53967ebd0f	func_curl: Add 'followlocation' option to CURLOPT() We allow for 'maxredirs' to be set, but this value is ignored when followlocation is not enabled which, by default, it is not. ASTERISK-17491 #close Reported by: candrews Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a	5 years ago
Sean Bright	b8963987a6	res_pjsip_endpoint_identifier_ip: Document support for hostnames ASTERISK-25429 #close Reported by: Joshua C. Colp Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824	6 years ago
Sean Bright	6648472a28	res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY ASTERISK-27775 #close Reported by: AvayaXAsterisk Change-Id: Iad158e908e34675ad98f74d09c5e73024e50c257	6 years ago
Friendly Automation	3e446508b4	Merge "netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13." into 13	6 years ago
Jaco Kroon	e7d2ab666b	netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr. ast_addressfamily_to_sockaddrize will determine the size that's required, and ast_sockaddr_from_sockaddr then wraps this new function and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without knowing the address family) into the ast_sockaddr structure. Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Kevin Harwell	fa700acf39	Merge "res_pjsip_pubsub: Add ability to persist generator state information." into 13	6 years ago
Joshua Colp	4cefb2f3cb	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 13	6 years ago
Joshua Colp	81899e71e7	Merge "app_agent_pool: Update XML docs for AgentLogin" into 13	6 years ago
Corey Farrell	7601a5f0f8	app_record: Do not hang up if beep audio is missing Additionally alter the warning to mention that it was "beep" which could not be streamed to give admins a better clue about what the warning means. ASTERISK-28682 Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e	6 years ago
Kevin Harwell	ba5b0c5c36	app_agent_pool: Update XML docs for AgentLogin This patch fixes some wrongly formatted documentation for the AgentLogin application. A couple of "see also" links should contain only the function name, and no parameters. Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661	6 years ago
George Joseph	83077366de	CI: Update buildAsterisk.sh to do a "make full" If you do a "make all" when building Asterisk the xml documentation produced will be missing certain AMI events where their documentation is located not at the top of the c source file but embedded further down next to the event's manager_event() registration call. See main/manager_mwi.c for an example. "make full" does produce the correct documentation so we're changing it in the build script. A separate commit/issue will address the problem with "make all". ASTERISK-28507 Reported by: David Lee Change-Id: I4a22635d6eef99eacecc0efb69e28360eebdb86c	6 years ago
Joshua Colp	379050504d	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic" into 13	6 years ago
Joshua C. Colp	9720b29d8e	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
Friendly Automation	1d3739fcbd	Merge "sig_pri: Fix deadlock caused by sig_pri_queue_hangup" into 13	6 years ago
Sean Bright	49bb7d85cd	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	aa84bc31e2	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 13	6 years ago
Friendly Automation	0b7569a977	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings" into 13	6 years ago
Friendly Automation	1577f6d923	Merge "features.c: Make Bridge application tolerate unspecified channel." into 13	6 years ago
Friendly Automation	b929ac2c64	Merge "app_dial.c: Simplify dialplan using Dial." into 13	6 years ago
Friendly Automation	beb7987fed	Merge "app_page.c: Simplify dialplan using Page." into 13	6 years ago
Friendly Automation	4db92ee97e	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message." into 13	6 years ago
Friendly Automation	3a86c176a6	Merge "app_chanspy.c: Reduce log message level from notice to verbose." into 13	6 years ago
George Joseph	e642d46b05	Merge "contrib/valgrind: Fix use of frame-level suppression" into 13	6 years ago
George Joseph	9249c30b87	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	3a8290d1f2	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	e4e5e07e3d	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	f6f25601a4	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	4414def9f9	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	9997710f8f	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	aed10616e5	app_chanspy.c: Reduce log message level from notice to verbose. Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	1cd2e340e0	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	430350ed53	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	0d252feab6	res_agi: Improve GET FULL VARIABLE documentation ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	6 years ago
Snuffy	c00c448c05	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
Friendly Automation	20c411065b	Merge "func_odbc: acf_odbc_read() and cli_odbc_read() unicode support" into 13	6 years ago
Joshua Colp	5a3826e334	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 13	6 years ago
Boris P. Korzun	d95b719891	func_odbc: acf_odbc_read() and cli_odbc_read() unicode support Added ast_odbc_ast_str_SQLGetData() considers SQL_DESC_OCTET_LENGTH column attribute for correct allocating the buffer. ASTERISK-28497 #close Change-Id: I50e86c8a277996f13d4a4b9b318ece0d60b279bf	6 years ago
George Joseph	7e66d2578e	Merge "db: Initialize condition primitive before use" into 13	6 years ago
Jean Aunis	72548ca9e5	chan_sip: voice frames are no longer transmitted after emitting a COLP The SIP transaction state was reset when emitting an UPDATE or a re-INVITE related to a COLP, preventing RTP packets to be emitted. ASTERISK-28647 Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b	6 years ago
Sean Bright	401056fd27	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
Friendly Automation	1a96383e29	Merge "config.c: Skip UTF-8 BOMs if present when reading config files" into 13	6 years ago
Jaco Kroon	e8ca2af820	netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13. This function is already present in 15, 16 and master. Change-Id: I491ddecdccc87620c264567c549217e48f3d4ff1 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago

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31270 Commits (bfc93cc95428532e89ac8ffd72084dc1c811229b) All Branches Search

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