asterisk

Commit Graph

Author	SHA1	Message	Date
Walter Doekes	d19dec53b5	chan_sip: Clarify in sample docs how directmediapermit/-acl should be used It said "restrict [...] which peers should be able to pass [audio] to each other". However, these settings are not global (for which you would expect signaling IPs to be checked). These settings are available per peer only, and the IPs being checked, are the RTP IPs. Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660	5 years ago
George Joseph	54cd865946	message.c: Add option to suppress the Message channel AMI and ARI events In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	5 years ago
Joshua Colp	3efeb147bb	Merge "res_pjsip_messaging: Allow Content-Type to be overridden" into 17	5 years ago
Friendly Automation	e7e65588e4	Merge "res_stasis: trigger cleanup after update" into 17	5 years ago
George Joseph	2c72e26478	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating." into 17	5 years ago
Friendly Automation	82eb8e7097	Merge "stasis/app: don't lock an app before a call to send" into 17	5 years ago
Friendly Automation	e86c32b878	Merge "res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly" into 17	5 years ago
Joshua C. Colp	35c9332edf	res_pjsip_pubsub: Increment persistence data ref when recreating. Each subscription needs to have a reference to the persisted data for it, as well as the main JSON contained within the tree. When recreating a subscription this did not occur and they both shared the same reference. ASTERISK-28714 Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a	5 years ago
Sean Bright	ef5702cef1	res_pjsip_messaging: Allow Content-Type to be overridden ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	5 years ago
Friendly Automation	5192a045f6	Merge "chan_sip: Always process updated SDP on media source change" into 17	5 years ago
Kevin Harwell	737bd8365e	stasis/app: don't lock an app before a call to send Calling 'app_send' eventually calls the app's message handler. It's possible for a handler to obtain a lock on another object, and then need/want to lock the app object. If the caller of 'app_send' locks the app object prior to calling then there's a potential for a deadlock, if another thread calls 'app_send' without locking. This patch makes it so 'app_send' is not called with the app object locked in the section of code doing such. ASTERISK-28423 #close Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27	5 years ago
Kevin Harwell	7031c3b7bd	res_stasis: trigger cleanup after update The cleanup code in stasis shuts down applications if they are in a deactivated state, and no longer have explicit subscriptions. When registering an app the cleanup code was running before calling 'update'. When it should be executed after 'update' since a call to register may re-activate the app. We don't want it to shutdown before the 'update' otherwise the app won't be re-activated, or registered. This patch makes it so the cleanup code is executed post 'update'. ASTERISK-28679 #close Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b	5 years ago
Sean Bright	69cf67d8c6	res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly We need to wait for the message sending callback to finish to know if we succeeded or failed. ASTERISK-25421 #close Reported by: Dmitriy Serov Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059	5 years ago
Walter Doekes	be6eec7b25	chan_sip: Always process updated SDP on media source change Fixes no-audio issues when the media source is changed and strictrtp is enabled (default). If the peer media source changes, the SDP session version also changes. If it is lower than the one we had stored, chan_sip would ignore it. This changeset keeps track of the remote media origin identifier, comparing that as well. If it changes, the session version needn't be higher for us to accept the SDP. Common scenario where this would've caused problems: a separate media gateway that informs the caller about premium rates before handing off the call to the final destination. (An alternative fix would be to set ignoresdpversion=yes on the peer.) ASTERISK-28686 Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee	5 years ago
Sean Bright	ace97a3786	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 check that the received frame is in a format we have negotiated 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
Asterisk Development Team	a0f44b1052	Update CHANGES and UPGRADE.txt for 17.2.0	5 years ago
Friendly Automation	6747dd31b4	Merge "http: Add ability to disable /httpstatus URI" into 17	5 years ago
George Joseph	f34a4d5a77	cdr.c: Set event time on party b when leaving a parking bridge When Alice calls Bob and Bob does a blind transfer to Charlie, Bob's bridge leave event generates a finalize on both the party_a and party_b CDRs but while the party_a CDR has the correct end time set from the event time, party_b's leg did not. This caused that CDR's end time to be equal to the answered time and resulted in a billsec of 0. * We now pass the bridge leave message event time to cdr_object_party_b_left_bridge_cb() and set it on that CDR before calling cdr_object_finalize() on it. NOTE: This issue affected transfers using chan_sip most of the time but also occasionally affected chan_pjsip probably due to message timing. ASTERISK-28677 Reported by: Maciej Michno Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca	5 years ago
Sean Bright	1e79bf44fc	http: Add ability to disable /httpstatus URI Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	5 years ago
Friendly Automation	f7e0b6baf8	Merge "func_odbc.conf.sample: Add example lookup" into 17	5 years ago
Friendly Automation	f556efe0b9	Merge "res_statsd: Document that res_statsd does nothing on its own" into 17	5 years ago
Friendly Automation	c8dc9365a4	Merge "translate.c: Fix silk 24kHz truncation in 'core show translation'" into 17	5 years ago
Joshua Colp	73cd259895	Merge "chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"" into 17	5 years ago
Friendly Automation	d6a4560016	Merge "chan_sip.c: Stop handling continuation lines after reading headers" into 17	5 years ago
Andrew Siplas	1c5116f842	chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout" The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX to match behavior of "no timeout" defined in comment. ASTERISK-28702 #close Change-Id: I4ea015986e061374385dba247b272f7aac60bf11	5 years ago
Sean Bright	687890f734	translate.c: Fix silk 24kHz truncation in 'core show translation' SILK @ 24kHz is not shown in the 'core show translation' output because of an off-by-one-error. Discovered while looking into ASTERISK~19871. ASTERISK-28706 Reported by: Sean Bright Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510	5 years ago
Sean Bright	d33cd39260	func_odbc.conf.sample: Add example lookup Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	5 years ago
Sean Bright	deed0f5706	res_statsd: Document that res_statsd does nothing on its own ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	5 years ago
Friendly Automation	503cac85ee	Merge "queue_log: Add alembic script for generate db table for queue_log" into 17	5 years ago
Joshua Colp	7a9a4575bc	Merge "app_voicemail, say: Fix various leading whitespace problems" into 17	5 years ago
Friendly Automation	88e3f5c315	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail" into 17	5 years ago
Rodrigo Ramírez Norambuena	7b32763190	queue_log: Add alembic script for generate db table for queue_log Change-Id: I35b928a6251f9da9a1742b2cd14c63a00c3d0f0c	5 years ago
Joshua Colp	68b7ef924e	Merge "pbx.c: Include filesystem cache in free memory calculation" into 17	5 years ago
Joshua Colp	cf37de8a5d	Merge "res_realtime: Fix 'realtime update2' argument handling" into 17	5 years ago
Joshua Colp	f132679193	Merge "app_voicemail: Set globals to default values when voicemail.conf missing" into 17	5 years ago
Sean Bright	cb6f106d76	app_voicemail, say: Fix various leading whitespace problems In `af90afd90c`, Japanese language support was added to app_voicemail and main/say.c, but the leading whitespace is not consistent with Asterisk coding guidelines. This patch fixes that. Whitespace only, no functional change. ASTERISK~23324 Reported by: Kevin McCoy Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87	5 years ago
Sean Bright	9ff062f994	pbx.c: Include filesystem cache in free memory calculation ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	5 years ago
Sean Bright	7c137ebbd4	chan_sip.c: Stop handling continuation lines after reading headers lws2sws() does not stop trying to handle header continuation lines even after all headers have been found. This is problematic if the first character of a SIP message body is a space or tab character, so we update to recognize the end of the message header. ASTERISK-28693 #close Reported by: Frank Matano Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df	5 years ago
Sean Bright	d1cd27ba1f	app_voicemail: Prevent crash when saving message with realtime voicemail ast_store_realtime() is not NULL tolerant, so we need to initialize the field values we pass to it to the empty string to avoid a crash. ASTERISK-23739 #close Reported by: Stas Kobzar Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c	5 years ago
Joshua Colp	8cad8db7d3	Merge "app_queue: Deprecate the QueueMemberPause.Reason field" into 17	5 years ago
Friendly Automation	af4e503005	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY" into 17	5 years ago
Sean Bright	b60dfac98e	app_voicemail: Set globals to default values when voicemail.conf missing If voicemail.conf exists but is empty, the config parsing process will default a number of global variables to non-zero values. On the other hand, if voicemail.conf is missing (arguably semantically equivalent to an empty file), this process is skipped and the globals are defaulted to 0. Set the globals to the same values they would be set to if a configuration were present. This allows voicemail configuration to be done completely by Realtime without the need to create an empty voicemail.conf file. ASTERISK-27622 #close Reported by: Jim Van Meggelen Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409	5 years ago
Sean Bright	62e5fa400e	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	0bd7cd562e	Merge "res_pjsip_endpoint_identifier_ip: Document support for hostnames" into 17	5 years ago
Friendly Automation	acbe620d41	Merge "func_curl: Add 'followlocation' option to CURLOPT()" into 17	5 years ago
Sean Bright	3779e76b68	res_realtime: Fix 'realtime update2' argument handling The change in `9b99ef50b5` updated the syntax of the 'realtime update2' CLI command but did not correctly update the calls to ast_update2_realtime(). The issue this addresses was originally opened because we aren't allowing a SQL NULL to be set as part of the update, but this is a limitation of the Realtime API and is not a bug. Additionally, this patch: * Corrects the example in the command documentation to reflect 'update2' instead of 'update.' * Fixes the leading spacing of the command documentation. * Checks that the required 'NULL' literal argument is present where we expect it to be. ASTERISK-21794 #close Reported by: Cédric Bassaget Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd	5 years ago
Joshua Colp	65bc8d70e3	Merge "app_record: Do not hang up if beep audio is missing" into 17	5 years ago
Sean Bright	a810025782	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	28275609b3	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	6898c8e0da	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

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32467 Commits (d19dec53b5c3eb01bcd288c6bac90fa35662fca0) All Branches Search

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