32748 Commits (fdc13060df8f79cd5277fb2c8a6c70d22b5cb69c)

 

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Author	SHA1	Message	Date
Sean Bright	dfad69ce7c	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	6 years ago
Sean Bright	262221f4d9	func_odbc.conf.sample: Add example lookup ... Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	6 years ago
Joshua Colp	dbb3a19c35	Merge "queue_log: Add alembic script for generate db table for queue_log"	6 years ago
Joshua Colp	64debbd13f	Merge "app_voicemail, say: Fix various leading whitespace problems"	6 years ago
Joshua Colp	058e9f735e	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail"	6 years ago
Joshua Colp	8208f5c73d	Merge "pbx.c: Include filesystem cache in free memory calculation"	6 years ago
George Joseph	0380288f7c	Merge "res_realtime: Fix 'realtime update2' argument handling"	6 years ago
Joshua Colp	2d17e25015	Merge "app_voicemail: Set globals to default values when voicemail.conf missing"	6 years ago
Sean Bright	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() ... * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit ec83b11183) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit fd64bb66f9) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	6 years ago
Sean Bright	5cbf47714a	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	6 years ago
Sean Bright	50d02d6194	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Sean Bright	f309b86e36	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	6 years ago
Sean Bright	ba8ccb9132	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	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Joshua Colp	d5fce4bc34	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY"	6 years ago
Friendly Automation	c665878e92	Merge "app_queue: Deprecate the QueueMemberPause.Reason field"	6 years ago
Sean Bright	9be89d9913	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	6 years ago
Friendly Automation	3098adc816	Merge "res_pjsip_endpoint_identifier_ip: Document support for hostnames"	6 years ago
Joshua Colp	be284af0d3	Merge "func_curl: Add 'followlocation' option to CURLOPT()"	6 years ago
Sean Bright	094e87b0dc	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	6 years ago
Friendly Automation	3f663a543d	Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr."	6 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. ... This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	6 years ago
Joshua Colp	f6f678fe7d	Merge "app_record: Do not hang up if beep audio is missing"	6 years ago
Sean Bright	0c2bf1664c	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	6 years ago
Sean Bright	9522390a69	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	6 years ago
Sean Bright	29d867ed67	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	90af050fa4	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	51f811183a	Merge "ARI: Ability to inhibit COLP frames when adding channels to a bridge"	6 years ago
Jaco Kroon	3bc8b36537	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
Friendly Automation	34746220a0	Merge "res_pjsip_pubsub: Add ability to persist generator state information."	6 years ago
Joshua Colp	a55d403429	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support"	6 years ago
Friendly Automation	9aaacdc7fc	Merge "CI: Update buildAsterisk.sh to do a "make full""	6 years ago
Corey Farrell	2f8b20b949	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	00a7432156	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	d5f3ec92d0	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 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