30374 Commits (4a461bcde45011ed1762b10273e38dcd2a3f2cab)

 

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Author	SHA1	Message	Date
Richard Mudgett	4a461bcde4	chan_pjsip.c: Improve ast_request() diagnostic msgs. ... Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and disable_multi_domain=no results in a misleading empty endpoint name message. The message should say the endpoint was not found. * Added missing endpoint not found message. * Added more information to the empty endpoint name msgs if available. * Eliminated RAII_VAR in request(). Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4	8 years ago
Jenkins2	eb23919e69	Merge "res_smdi: Fix shutdown ref."	8 years ago
Jenkins2	c9bcd888a2	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams."	8 years ago
Jenkins2	6a0505eee0	Merge "res_hep: hepv3_is_loaded() should check if we are enabled"	8 years ago
Joshua Colp	73bd9d6488	Merge "res_clialiases: Fix completion pass-through."	8 years ago
Jenkins2	26e8de9453	Merge "coverity: Fix warnings in res_smdi"	8 years ago
Joshua Colp	b223f0e108	Merge "loader: Minor fix to module registration."	8 years ago
Jenkins2	bcb4e6e608	Merge "res_musiconhold: Start playlist after initial announcement"	8 years ago
Joshua Colp	b0f054cb2d	Merge "app_queue: Fix extension state subscriptions removed on dialplan reload"	8 years ago
Jenkins2	dff0415b1e	Merge "pjsip_options: wrongly applied "UNKNOWN" status"	8 years ago
Corey Farrell	03c25a869f	res_smdi: Fix shutdown ref. ... When adding shutdown refs for OPTIONAL_API components I accidentally added it to the unload_module function in res_smdi. Move it to load_module. Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c	8 years ago
Sean Bright	9755eff46f	res_hep: hepv3_is_loaded() should check if we are enabled ... res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of unnecessary work even if 'enabled' is set to 'no' in hep.conf. Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b	8 years ago
Jenkins2	e7a6e64039	Merge "pjsip: Ignore state changes from old transactions."	8 years ago
Corey Farrell	80bf0ee99a	loader: Minor fix to module registration. ... This protects the module loader itself against crashing if dlopen is called on a module from outside loader.c. * Expand scope of lock inside ast_module_register to include reading of resource_being_loaded. * NULL check resource_being_loaded. * Set resource_being_loaded NULL as soon as dlopen returns. This fixes some error paths where it was not NULL'ed. * Create module_destroy function to deduplicate code from ast_module_unregister and modules_shutdown. * Resolve leak that occured if a module did not successfully register. * Simplify checking for successful registration. Change-Id: I40f07a315e55b92df4fc7faf525ed6d4f396e7d2	8 years ago
Jenkins2	30658e7143	Merge "configs: Comment out and change IP of iax.conf [demo]"	8 years ago
Corey Farrell	a8aa209901	res_clialiases: Fix completion pass-through. ... Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8	8 years ago
Jenkins2	a33207a91f	Merge "res_pjsip_session: Reinvite using active stream topology if none requested."	8 years ago
Richard Mudgett	98f7e9251f	res_rtp_asterisk.c: Disable packet flood detection for video streams. ... We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
George Joseph	283d2df680	res_pjsip_sdp_rtp: Add NULL check in add_crypto_to_stream ... add_crypto_to_stream wasn't checking for a NULL session->inv_session->neg before calling pjmedia_sdp_neg_get_state. This was causing a crash if the negotiation hadn't already been completed and asterisk was compiled with --enable-dev-mode. Change-Id: I57c6229954a38145da9810fc18657bfcc4d9d0c9	8 years ago
Sean Bright	c387beb456	res_musiconhold: Start playlist after initial announcement ... Reset the samples counter to zero when we are done playing an announcement so that we don't skip into the middle of the first file in the playlist. Also add the selected annoucement to the output of 'moh show classes.' ASTERISK-24329 #close Reported by: Thomas Frederiksen Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0	8 years ago
Sean Bright	7a8a187a56	coverity: Fix warnings in res_smdi ... ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a	8 years ago
Sean Bright	dac5e3a0df	configs: Comment out and change IP of iax.conf [demo] ... This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100	8 years ago
Joshua Colp	ec984870b0	Merge "menuselect: Tweak check for recently run configure."	8 years ago
Joshua Colp	49f2ff37bd	Merge "README: Remove outdated references to tex docs"	8 years ago
Joshua Colp	7a837ece97	Merge "Add new AMI action for app_voicemail"	8 years ago
Joshua Colp	e411b7d112	Merge "chan_sip: 3PCC patch for AMI "SIPnotify""	8 years ago
Kevin Harwell	30954337a0	Merge "pjsip_options: contacts sometimes not being updated on reload"	8 years ago
Jenkins2	588be919cb	Merge "res_pjsip: Assign support levels to a few modules"	8 years ago
George Joseph	a51bfe5a79	README: Remove outdated references to tex docs ... Added links to the wiki to replace references to outdated tex docs. ASTERISK-27430 Reported by: Corey Farrell Change-Id: I5007e732b30bc7b63d124c530ae8857c89991209	8 years ago
Jenkins2	8a281776d5	Merge "pjsip_options: dynamic contact's fields not updated on reload"	8 years ago
George Joseph	afb6e85980	Merge "CLI: Fix 'core show sysinfo' function ordering."	8 years ago
Joshua Colp	c50905756b	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"	8 years ago
pchero	daa3a3009a	Add new AMI action for app_voicemail ... Currently, to figure out specified voicemail's status, there's only one way to do it, which is use a VoicemailUserEntry AMI message. But it consumed it too much resource(it check everything). So, added new AMI action. ASTERISK-27470 Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7	8 years ago
Joshua Colp	62f2860c39	AST-2017-012: Place single RTCP report block at beginning of report. ... When the RTCP code was transitioned over to Stasis a code change was made to keep track of how many reports are present. This count controlled where report blocks were placed in the RTCP report. If a compound RTCP packet was received this logic would incorrectly place a report block in the wrong location resulting in a write to an invalid location. This change removes this counting logic and always places the report block at the first position. If in the future multiple reports are supported the logic can be extended but for now keeping a count serves no purpose. ASTERISK-27382 ASTERISK-27429 Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116	8 years ago
Jenkins2	58514c7442	Merge "chan_sip: Don't crash in Dial on invalid destination"	8 years ago
Joshua Colp	3370cd21df	res_pjsip_session: Reinvite using active stream topology if none requested. ... When a connected line update is sent to an endpoint we do not request a specific stream topology to be used. Previously this resulted in the configured stream topology being used which may actually differ from the currently negotiated topology. PJSIP is helpful in this regard in that it will fill in any missing streams with removed ones. This results in our own state not matching the SDP, though, and we do not apply the negotiated SDP. This change tweaks the code to use the actively negotiated stream topology if it is present with a fallback to the configured one. This results in the SDP and the state having matching information and the world is happy. ASTERISK*27397 Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604	8 years ago
Jenkins2	e7dccbe708	Merge "chan_sip: Don't send trailing \0 on keep alive packets"	8 years ago
Joshua Colp	0b532367bd	pjsip: Ignore state changes from old transactions. ... When we fail over to a new target we create a new transaction and it becomes the current INVITE transaction. This does not prevent the previous transaction from raising state changes and causing the session to be prematurely disconnected if a transport error occurs immediately. This change backports a fix from PJSIP that eliminates the incorrect state change and reduces when they would be raised in the first place. ASTERISK-27408 Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34	8 years ago
Yasuhiko Kamata	cb249b2419	chan_sip: 3PCC patch for AMI "SIPnotify" ... A patch for sending in-dialog SIP NOTIFY message with "SIPnotify" AMI action. ASTERISK-27461 Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4	8 years ago
Ivan Poddubny	c7f94e570e	app_queue: Fix extension state subscriptions removed on dialplan reload ... The approach with having a single global subscription to all extension state changes has one issue: dynamically created hints don't have any watchers and are therefore garbage collected on the first dialplan reload. This change creates a state subscription for every queue member with a hint as state_interface, thus increasing the count of watches for hints, so they are not destroyed prematurely anymore. There are 2 side effects: 1. The state change callback in app_queue is not executed when there are no members referring to the extension. 2. The callback is called multiple times for the same hint if it's associated with more than one queue member. Reported by: Steven T. Wheeler ASTERISK-18411 #close Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89	8 years ago
Sean Bright	0c9cc7e975	chan_sip: Don't send trailing \0 on keep alive packets ... This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420	8 years ago
Sean Bright	5039b5741c	chan_sip: Don't crash in Dial on invalid destination ... Stripping the DNID in a SIP dial string can result in attempting to call the argument parsing macros on an empty string, causing a crash. ASTERISK-26131 #close Reported by: Dwayne Hubbard Patches: dw-asterisk-master-dnid-crash.patch (license #6257) patch uploaded by Dwayne Hubbard Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e	8 years ago
Corey Farrell	6a67828b46	menuselect: Tweak check for recently run configure. ... Recently menuselect has randomly produced an error stating that configure was just run and make had to be restarted. I believe this is due to an incorrect menuselect/Makefile rule. The original rule produced an error if makeopts or autoconfig.h were older than makeopts.in or autoconfig.h.in. I believe this can create an issue if makeopts is older than autoconfig.h.in or if autoconfig.h is older than makeopts.in. The new rules compare files independently. Change-Id: Ibca155035fa1392c95e33cbf25f257902abba17b	8 years ago
Richard Mudgett	22810fc635	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) ... This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Sean Bright	ec1f4bf48d	res_pjsip: Add TLSv1.1 and TLSv1.2 support ... Support for these protocols was added in the same commit as the 'proto' field, so we can safely use the same ./configure check. For reference: https://trac.pjsip.org/repos/changeset/4968 Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac	8 years ago
Sean Bright	0b9d2135a9	res_pjsip: Assign support levels to a few modules ... Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d	8 years ago
Corey Farrell	c01ba7437e	CLI: Fix 'core show sysinfo' function ordering. ... Handle CLI initialization before any processing occurs. Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c	8 years ago
Kevin Harwell	b088cddc03	pjsip_options: wrongly applied "UNKNOWN" status ... A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89	8 years ago
Richard Mudgett	c2ec82bf36	stasis_channels.c: Don't set channel snapshot caller_dnid twice. ... Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471	8 years ago
Jenkins2	38dcdf2d68	Merge "codec_opus: Make libcurl a dependency in menuselect"	8 years ago