asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	f5d4e581f3	apps/app_dial: Fix Dial 'z' option The 'z' option is supposed to disable the dial timeout in the case of a call forward. Unfortunately, the wrong timeout timer was passed to the do_forward function, resulting in the option not working. ASTERISK-24225 #close Reported by: dimitripietro Tested by: dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621) ........ Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	524cb990a3	configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is executed with optimization. This "help" unfortunately results in re-definition warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning. Review: https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close Reported by: Kilburn Tested by: Kilburn, wdoekes patches: 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by cloos (License 5956) 11.diff uploaded by cloos (License 5956) 12.diff uploaded by cloos (License 5956) 13.diff uploaded by cloos (License 5956) ........ Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	66fb08e26d	res_http_websocket: Include query parameters in client connection requests. Review: https://reviewboard.asterisk.org/r/3914/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2c013ae774	Bridging: Fix a behavioral change when checking if a channel is leaving a bridge r420934 introduced some failures in the test suite. Upon investigating, it was discovered that differences in the way we were evaluating whether a channel was in the process of leaving a bridge were causing some reinvites not to occur (mostly reinvites back to Asterisk when ending a call). This patch fixes that behavioral change. ASTERISK-24027 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3910/ ........ Merged revisions 421186 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	544e092b2d	app_voicemail/app: Remove test events that were duplicated by r421059 Moving the test event raised when a file is played back (which occurred in r421059) broke the ever loving snot out of the voicemail tests. This caused duplicate test events to get raised, as app_voicemail and main/app were raising events prior to call ast_streamfile. The voicemail tests did not enjoy getting multiple events. Since raising the playback event in ast_streamfile is far more useful to the vast majority of tests, this patch keeps the call there and simply removes the extraneous calls that duplicated the event. ........ Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cce3d9ec5c	res/res_hep_rtcp: Remove dependency on PJSIP The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need to be included, as the module does not using PJPROJECT any fashion. Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as a dependency, this also meant that res_hep_rtcp will fail to compile on a system without PJPROJECT. This patch removes the include. Thanks to Damien Wedhorn for pointing this out in #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn, Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fa02e06132	main/file: Move test event to emit PLAYBACK event more consistently This is being done in advance of the test for ASTERISK-23953 ........ Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421060 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421061 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6e4d44c2a1	cel: Make sure channels in extra fields include their unique IDs as well CEL typically tracks a lot of information using the unique ID of the channel. This is typically needed due to tying events together using the linked ID of the various channels involved in a "call", which is derived from the channel ID of the oldest channel involved in a bridge (or in the case of a Dial, the parent channel). Previously, we had updated the extra fields to include the involved channel names, but forgot to put in the unique ID. This patch corrects that error. ........ Merged revisions 421037 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ee93b5a314	ARI: Originate to app local channel subscription code optimization. Reduce the scope of local_peer and only get it if the ARI originate is subscribing to the channels. Review: https://reviewboard.asterisk.org/r/3905/ ........ Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7eb4ee9b2f	channel_internal_api.c: Replace some code with ao2_replace(). Use ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace() has the advantange of not altering the ref count if the replaced pointer is the same. Review: https://reviewboard.asterisk.org/r/3904/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cd81f920a4	res_pjsip_send_to_voicemail.c: Fix svn file properties. ........ Merged revisions 420956 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e8a5847742	PJSIP: Prevent crash no-URI contacts This prevents a crash from occurring when a contact with no URI is used for the creation of an outbound out-of-dialog request with no associated endpoint. ........ Merged revisions 420949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	cd28e5dda2	Bridges: Fix feature interruption/unintended kick caused by external actions If a manager or CLI user attached a mixmonitor to a call running a dynamic bridge feature while in a bridge, the feature would be interrupted and the channel would be forcibly kicked out of the bridge (usually ending the call during a simple 1 to 1 call). This would also occur during any similar action that could set the unbridge soft hangup flag, so the fix for this was to remove unbridge from the soft hangup flags and make it a separate thing all together. ASTERISK-24027 #close Reported by: mjordan Review: https://reviewboard.asterisk.org/r/3900/ ........ Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e6022f9f97	AMI: Improve documentation for Status action git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	602aef327e	logger: Don't store verbose-magic in the log files. In r399267, the verbose2magic stuff was edited. This time it results in magic characters in the log files for multiline messages. In trunk (and 13) this was fixed by the "stripping" of those characters from multiline messages (in r414798). This fix is altered to actually strip the characters and not replace them with blanks. Review: https://reviewboard.asterisk.org/r/3901/ Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged revisions 420897 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420898 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f0a65379f5	chan_sip: Fix type mismatch when the format is changed. Symptom is most likely an invalid ao2 object bad magic number message or a less likely crash. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bede29b762	res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup. * Made use ast_copy_string() instead of strcpy() for snoop uniqueid for safety. There is no guarantee that the max channel uniqueid length will remain the same as the snoop uniqueid space. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a2bbe5d360	app_voicemail: Fix the "test_voicemail_vm_info" unit test. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a0b7f2ce42	res/stasis/command.c: Fix recent commit using spaces instead of tabs. ........ Merged revisions 420836 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e30904854e	AMI/ARI: Update version to 2.5.0/1.5.0 respectively This is to support the backwards compatible changes made in the next version of Asterisk. ........ Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	ccb2f94691	Stasis: Use the correct return value Return the correct value instead of always returning 0 when setting internal status on unreal channels. Reported by: Richard Mudgett ........ Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	406dded64c	Stasis: Allow internal channels directly into bridges The patch to catch channels being shoehorned into Stasis() via external mechanisms also happens to catch Announcer and Recorder channels because they aren't known to be stasis-controlled channels in the usual sense. This marks those channels as Stasis()-internal channels and allows them directly into bridges. Review: https://reviewboard.asterisk.org/r/3903/ ........ Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	ef70c08dc7	Improve call forwarding reporting, especially with regards to ARI. This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1b500d2fa1	Fix crashing unit tests with regards to RLS. The unit tests require a sorcery.conf file that has been set up to store resource lists in memory rather than retrieving from configuration. With a setup that is not conducive to running the tests, a fault in sorcery currently causes Asterisk to crash when attempting to run any of the tests. To get around the crash, this adds a function that verifies the current environment and marks the tests as "not run" if the setup is not correct. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c43c22fe89	Fix crash encountered by the testsuite. Running testsuite tests locally produced no errors, but when run using the continuous integration framework, crashes occurred. The crashes occurred due to a refcounting error that had been fixed for a similar situation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc7853f40f	res_hep: Remove disabling of modules These modules were originally specified as being disabled, as they were introduced midstream in Asterisk 12. That makes it nicer for folks who are upgrading to a new release in the middle of Asterisk 12. That's not the case for Asterisk 13: it's a brand new release. There's no reason to have the modules disabled by default in that case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	b94fc06966	general: Fix memory Corruption in __ast_string_field_ptr_build_va. If the space left in a stringfield is between 0 and (alignof(ast_string_field_allocation)-1) adding new data would cause memory corruption, because we would assume enough space (unsigned underrun). Thanks Arnd Schmitter for reporting and finding out the cause! ASTERISK-23508 #close Reported by: Arnd Schmitter Tested by: Arnd Schmitter, JoshE Review: https://reviewboard.asterisk.org/r/3898/ ........ Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	4e07345c28	tcptls: Avoid compiler warning on non-dev-mode. ........ Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6c34de22b5	funcs/func_jitterbuffer: Tweak documentation This patch merely reformats and cleans up a bit of the jitterbuffer documentation for the wiki. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	95871451f6	app_queue: Add RealTime support for queue rules This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8b411f710b	Update CHANGES file git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	af6f95f64c	Update UPGRADE.txt file git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	7438e38f22	Fix build in devmode. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	2b804661bc	app_voicemail: Add the ability to specify multiple email addresses. ASTERISK-24045 Reported by: Jacob Barber Review: https://reviewboard.asterisk.org/r/3833/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	02fc8e2449	chan_sip: Mark chan_sip and its files as extended support git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fbb612751e	make_ari_stubs: Update wiki prefix to '13' git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5b82edd783	res_ari_resource.c.mustache: Update template to emit module support level git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	949c59f25f	main/message: remove debug message ........ Merged revisions 420533 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d2dff48f70	CEL: Update unit tests for additional information This updates the CEL unit tests for the new information contained in the attended transfer CEL extra field. ........ Merged revisions 420513 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8b560be831	Update UPGRADE file for 13 branch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	288f57882e	Remove old properties git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d4f4162258	___ _ _ _ __ _____ / _ \ \| \| (_) \| \| / \|\|____ \| / /_\ \___\| \|_ ___ _ __ _ ___\| \| __ `\| \| / / \| _ / __\| __/ _ \| '__\| / __\| \|/ / \| \| \ \ \| \| \| \__ \| \|\| __\| \| \| \__ \| < _\| \|.___/ / \_\| \|_\|___/\__\___\|_\| \|_\|___\|_\|\_\ \___\____/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a1424a2f1a	chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue. Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ ........ Merged revisions 420434 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420435 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420436 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6653d7e8ee	Stasis: Correct blind transfer message generation This fixes the json object creation format string and key name for the BridgeBlindTransfer Stasis event allowing it to be published properly. ........ Merged revisions 420414 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d442ffd59c	Stasis: Ensure transfer messages follow validation rules This makes Stasis() event generation for transfer messages follow validation rules. Currently, ast_json_null() is being used in place of omitting a key entirely which falls afoul of these validation rules. https://reviewboard.asterisk.org/r/3892/ ........ Merged revisions 420408 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	965ba7c36f	Fix build in dev mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	cb55679aed	Ensure bridges exist when trying to determine bridged parties when publishing transfer information. ........ Merged revisions 420387 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	99d0bccd35	Add support for RFC 4662 resource list subscriptions. This commit adds the ability for a user to configure a resource list in pjsip.conf. Subscribing to this list simultaneously subscribes the subscriber to all resources listed. This has the potential to reduce the amount of SIP traffic when loads of subscribers on a system attempt to subscribe to each others' states. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ea7d4ab09e	chan_iax2: Several media format fixes. * Fixed the iax.conf bandwidth option. This is the root cause of ASTERISK-24150. * Added checks in iax2_request() to ensure that there are actual formats requested for the new channel to prevent any more fracks from issues like ASTERISK-24150. This is a consequence of the iax.conf bandwidth option not working. * Fixed struct iax2_codec_pref.order member size mismatch issue when converting to and from the codec preference order list passed over the wire. In addition the values sent over the wire are now compatible with previous Asterisk versions. * Fixed several issues dealing with the struct iax2_codec_pref members. Off-by-one, array limit errors, and the order/framing members always need to be updated together. * Made iax2_request() setup the channel's native format preference order according to the user's wishes. The new media format strategy needs the order specified earler. * Fixed usage of ast_format_compatibility_bitfield2format(). The function can return NULL if the bitfield was not associated with a function. * Deleted dead code iax2_codec_pref_getsize() and iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8. * Renamed prefs to prefs_global so it won't get confused with the local pref versions. * Fixed too small buffer in handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete lines. * Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an optimization so iax2_request() and iax2_call() do less work. * Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when the pbx could not get started. * Made set_config() setup a local prefs list along side the local capability format bitfield. Once the config is loaded, then the local copies are put into the global versions. * Fix unininialized codec_buf in function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3890/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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