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Author	SHA1	Message	Date
Matthew Jordan	cb9ae40a31	test_cel: Update pickup test to expect CANCEL instead of ANSWSER ... The CEL pickup test previously looked for a disposition of ANSWER between the original caller/peer when the call is picked up. This is actually incorrect: the disposition should, at the very least, not be ANSWER as the call was never ANSWERed. The disposition is now CANCEL; this patch updates the test accordingly. ........ Merged revisions 425757 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fdcec1ef40	main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size' ... When refactoring CDRs to use the configuration framework, a 'whoops' was introduced where the CDR batch size was used when rescheduling a batch, as opposed to the time duration. This patch corrects that obvious mistake. ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged revisions 425735 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b8f505294a	config: Fix inf loop using ast_category_browse and ast_variable_retrieve ... Fix infinite loop when calling ast_variable_retrieve inside an ast_category_browse loop when there is more than 1 category with the same name. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4089/ ........ Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	289830cdc6	PJSIP: Enforce module load dependencies ... This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	50e802445c	Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379. ...	11 years ago
Joshua Colp	70a3e74f0a	res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't. ... In the case where the ICE negotiation had not yet started current state would get wiped when it shouldn't. This also removes channel binding as in practice this does not work well with other implementations. ........ Merged revisions 425644 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425645 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1f1a352fbd	chan_motif: Cleanup jingle_tech.capabilities only once. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	fab0df2f4c	parking_tests: Fix assertions and possibly crashes in res_parking unit tests ... Assertions were caused by attempting to play music on hold to a channel with no formats. Parking unit test channels were given formats and a technology so that they would be able to pretend to read/write frames. ASTERISK-24413 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4075/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	edf5af72e9	chan_ooh323: fix rtptimeout general value checking ... correct condition to check rtptimeout in [general] config section ASTERISK-24393 #close Reported by: Dmitry Melekhov Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........ Merged revisions 425547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425548 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	54a5b8798f	config: Fix SEGV in unit test with MALLOC_DEBUG ... With MALLOC_DEBUG the /main/config config_basic_ops test was causing a SEGV while doing an ast_category_delete in an ast_category_browse loop. Apparently this never worked but was also never tested. I removed the test, added 2 notes to config.h indicating that it's not supported and added a few lines of code to ast_category_delete to prevent the SEGV should someone attempt it in the future. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4078/ ........ Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2d07283dc5	Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute ... Tasks that were marked for pending deletion in the scheduler would be moved to the cache for later reuse, but after being recycled the deleted mark wouldn't be removed resulting in fresh tasks being deleted without reason... and immediately moved back into the cache where they could be reused again. This could cause horrendous things to happen in just about anything that used a scheduler. ASTERISK-24321 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4071/ ........ Merged revisions 425503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	26724c65f7	res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup ... Based on feedback from Richard, I created an accessor for res_phoneprov/ast_phoneprov_std_variable_lookup and added load priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ ........ Merged revisions 425480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c0943cf8d6	res_fax: Fix reference leak caused by gateway sessions ... Fax gateway session objects can be re-used, causing the same gateway session to be added to faxregistry.container more than once. This change causes fax_session_new to remove the reserved session from the container before it's id is changed, ensuring it's possible for the session to be freed. ASTERISK-24392 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4049/ ........ Merged revisions 425457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425459 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9a7fb8ee6c	stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2) ... Masquerades into and out of channels that are involved in a dial operation don't create the expected dial end event. The missing dial end event goes against the model for things like CDRs and generating Dial end manager actions and such. There are four cases: 1) A channel masquerades into the caller channel. The case happens when performing a blonde transfer using the channel driver's protocol. 2) A channel masquerades into a callee channel. The case happens when performing a directed call pickup. 3) The caller channel masquerades out of dial. The case happens when using the Bridge application on the caller channel. 4) A callee channel masquerades out of dial. The case happens when using the Bridge application on a peer channel. As it turned out, all four cases need to be handled instead of just the first one. ASTERISK-24237 Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ ........ Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	6b93ab5824	res_fax: Resolve module reference leak caused by reserved sessions ... Remove reference to module providing reserved session after adding a reference to the final module. This re-reference is done to ensure that module references are correct even if the final session selects a different module than the reserved session. ASTERISK-18923 #close Reported by: Grigoriy Puzankin Review: https://reviewboard.asterisk.org/r/4048/ ........ Merged revisions 425405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425407 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425411 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8bb06d592a	manager/config: Support templates and non-unique category names via AMI ... This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	efefbe15a4	res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'. ... ........ Merged revisions 425360 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425361 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	0ebe3d78bc	chan_sip: Fix so asterisk won't send reINVITE after a BYE. ... After a reINVITE glare situation, Asterisk would re-send the reINVITE even though the call had been hung up in the mean time. This patch unschedules the reinvite when handling the BYE. ASTERISK-22791 #close Reported by: Paolo Compagnini Tested by: Paolo Compagnini Review: https://reviewboard.asterisk.org/r/4056/ (testcase is in review r4055) ........ Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425297 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425298 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	6644fd46e7	build: Relax badshell tilde test to allow for ~ in middle of DESTDIR. ... The main Makefile has a target test called 'badshell' that tests if DESTDIR does not happen to have an an-expanded tilde (~). This might be the case if you run: make install DESTDIR=~/somewhere/ That test also disallowed valid tildes in directory names. The test is now changed to only trigger on a tilde at the start of the path. ASTERISK-13797 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4064/ ........ Merged revisions 425291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425292 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425293 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	1e6c3d1a26	res_calendar_ews: Relax neon version check to work with 0.30 too. ... Allow res_calendar_ews to work not only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4068/ ........ Merged revisions 425286 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425287 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425288 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	004c387041	res_phoneprov: Cleanup module load error handling ... Tested module load/reload interaction between res_phoneprov and res_pjsip_phoneprov_provider in cases where res_phoneprov didn't load correctly (usually misconfiguration or missing phoneprov.conf) Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4069/ ........ Merged revisions 425264 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	743ad19699	bridge: During a smart bridge operation provide a more complete bridge to the old technology. ... When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b8aed5b14d	res/res_phoneprov: Bail on registration if res_phoneprov didn't load ... If res_phoneprov failed to fully load (due to not being configured), the providers container will be NULL. If a module attempts to register a phone provisioning provider, it should check for the presence of the container. If there is no providers container, it should return an error. This patch makes the ast_phoneprov_provider_register function do that... otherwise this would be a silly commit message. ........ Merged revisions 425220 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	23ffd68e70	res_pjsip_phoneprov_provider: Add missing dependency on pjproject. ... ........ Merged revisions 425216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	32624fb541	CallerID: Fix parsing regression ... This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	494bb9f931	res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on. ... When the 'force_rport' option is enabled the behavior should be the same as if the remote side placed rport into the message themselves. Therefore any responses we send should include the source port of the request in the rport of the Via header. #SIPit31 ASTERISK-24387 #close Reported by: Matt Jordan ........ Merged revisions 425131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	4c2aef333c	chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE. ... If a device re-INVITEs at the same time as the dialog is hung up, and if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would fail to destroy the dialog after a while. This resulted in (most prominently) file handle leaks. (Patch reindented by me.) ASTERISK-20784 #close ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) Reviewboard: https://reviewboard.asterisk.org/r/4052/ (testcase can be found at r4051) ........ Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425069 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	bbc56596fd	res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR ... endpoint->inbound_auths was changed to a vector in 13 and I committed the 12 patch instead of the 13 patch. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	e2ae7bd79f	res_rtp_asterisk: Crash if no candidates received for component ... When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ ........ Merged revisions 425030 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d3342fb98d	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov ... This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc07f4835c	res/res_phoneprov: Don't cancel Asterisk load on module load failure ... ........ Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d1c9621852	res_phoneprov: Refactor phoneprov to allow pluggable config providers ... This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	376ec31b17	cdr.c: Make turning on CDR debug a one step process instead of two. ... Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	4c7887b852	safe_asterisk: Don't automatically exceed MAXFILES value of 2^20. ... On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can exceed the per-process file limit of 2^20. This patch ensures the value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) ........ Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424878 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424879 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0292839ae0	res_rtp_asterisk: Allow only UDP ICE candidates. ... The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp ........ Merged revisions 424852 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424853 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b32d8b5317	Stasis: Relegate log message to dev-mode ... This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	9e08180304	Indexer: Format message types may not exist ... In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	57a5e2ebee	Stasis: Only log errors for non-declined types ... When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	673c2febba	data: Properly access formats in capabilities structure when adding codecs. ... Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	17d079c371	res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter ... Prior to this patch, the auth_reject_permanent parameter was not initialized on the registration client state, leading to the parameter being disabled regardless of the value specified in pjsip.conf. This patch initialized the setting on the registration client state to the provided configuration value. ASTERISK-24398 #close ........ Merged revisions 424730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9f5c73586c	res/res_pjsip_pubsub: Fix typo in WARNING message ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5c607f9735	message: Don't close an AMI connection on SendMessage action error ... If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6a8cb946eb	features.c: Fix lingering channel ref while Bridge() application is active. ... Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c092e49344	sdp_srtp: Add new lines to some WARNING messages ... ........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4f28ae4f51	res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint ... An OPTIONS request that is sent to Asterisk but not to a specific endpoint is currently sent a 404 in response. This is because, not surprisingly, an empty extension is never going to be found in the dialplan. This patch makes it so that we only attempt to look up the endpoint in the dialplan if it is specified in the OPTIONS request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt Jordan ........ Merged revisions 424624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	57233a97e8	pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels ... Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health. It will treat the channels as a PJSIP channel, eventually hitting an ao2 error, FRACKing on assertion error, and quite likely crashing. This patch adds checks to the read/write callbacks that ensure that the channel technology is of type 'PJSIP' before attempting to operate on the channel. #SIPit31 ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	69032d62aa	res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message ... When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f36b64f58e	res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions ... A subscription that has been persisted can - for various reasons - fail to be re-created on startup. This patch resolves a number of crashes that occurred when a subscription cannot be re-created on several off-nominal paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9e3b5be182	Release AMI connections on shutdown. ... ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0904b18fcc	Blocked revisions 424575 ... ........ chan_sip: Clean leak on error path of process_sdp Resolve leak in process_sdp that occurs in 2 error path's where crypto lines are expected but not provided. ASTERISK-24385 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4045/ ........ Merged revisions 424569 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago