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Author	SHA1	Message	Date
Joshua Colp	0de2d080c2	res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session. ... #SIPit31 ........ Merged revisions 424287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e3da76a352	PJSIP: Handle defaults properly ... This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	ac095304e6	PJSIP: Force transport on contact rewrite ... If contact rewriting is enabled but the contact differs in transport from what is actually being used, messages after the initial INVITE transaction can be sent to an incorrect transport/port combination. In the case where this bug occurred the remote party never received a BYE since it was sent to the remote party's TCP port over UDP. Review: https://reviewboard.asterisk.org/r/4032/ ........ Merged revisions 424244 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	303547231e	chan_sip: Simplify some unref code by removing unlink_peer_from_tables. ... ASTERISK-22945 #related Reported by: ibercom Patches: asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) ........ Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424182 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424183 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	45e32e4b8c	chan_sip: Remove excess ref of realtime peer before sip_poke_peer. ... The peer is referenced at the end of sip_poke_peer, it should not get an extra ref before the call to sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: asterisk11.patch uploaded by ibercom (License #6599) Review: https://reviewboard.asterisk.org/r/4031/ ........ Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424177 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424178 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d9b15388b2	res_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6 addresses in brackets. ... #SIPit31 ........ Merged revisions 424155 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b1bb6b97df	res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. ... This change fixes an issue where ICE candidates put into the SDP did not contain the 'raddr' and 'rport' information for server reflexive and relay candidates. #SIPit31 ........ Merged revisions 424151 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	faae530006	pjsip_cli: Suppress header print on error or no objects ... If there's an error on the pjsip command line or there are no objects, don't print the column headers. ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George Joseph Tested-by: Brad Latus Review: https://reviewboard.asterisk.org/r/4025/ ........ Merged revisions 424128 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8d55892df7	autosupport: Fix bashism. ... '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a 'case' works better there. Originally committed in r375059 and r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported by: Tzafrir Cohen ........ Merged revisions 424117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424125 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2a7c5208ee	Simplify UUID generation in several places. ... Replace code using ast_uuid_generate() with simpler and faster code using ast_uuid_generate_str(). The new code avoids a malloc(), free(), and copy. ........ Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	00207158e1	threadpool.c: Minor cleanup fixes. ... * Fix threadpool_alloc() prototype. * Add missing off-nominal NULL check of pool in threadpool_alloc(). * searializer_create() does not need to create the object with a lock as the lock is not used. ........ Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	19ffbb1e64	res_pjsip_session: Add additional checks for delaying session refreshes. ... There are certain situations which no checks existed for which need to prevent session refreshes. This includes sending a session refresh with SDP before SDP negotiation has completed and sending a session refresh before the dialog itself has been established. Checks for these have been added. Additionally COLP related UPDATEs were including SDP when it is not needed. Review: https://reviewboard.asterisk.org/r/4008/ ........ Merged revisions 424056 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5a77eb3476	res_fax: Fix out of bounds error in update_modem_bits(). ... ASTERISK-24357 #close Reported by: Jeremy Laine Patches: res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine Modified patch to not use magic numbers. ........ Merged revisions 423979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423983 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423987 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	77de3be28d	docs: Escape unescaped minus sign in asterisk.8 manpage. ... ASTERISK-23768 #close Reported by: Jeremy Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561) ........ Merged revisions 423915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423916 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423917 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8ae471258e	res_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous(). ... * Made memset the std struct in ast_sip_push_task_synchronous() because if DEBUG_THREADS is enabled then uninitialized lock tracking data is used. ........ Merged revisions 423894 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	774890de1b	pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request. ... The crash on the issues is a result of an invalid transport configuration change when asterisk is restarted. The attempt to send the qualify request fails and we cleaned up. However, the callback is also called which results in a double unref of the objects involved. * Put a wrapper around pjsip_endpt_send_request() to detect when the passed in callback is called because of an error so callers can know to not cleanup. * Made send_request_cb() able to handle repeated challenges (Up to 10). * Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding it. The sched entry will no longer self stop and must be externally stopped. * Added REF_DEBUG description tags to struct sched_data in pjsip_options.c. * Fix some off-nominal ref leaks in schedule_qualify(), qualify_and_schedule(). * Reordered pjsip_options.c module start/stop code to cleanup better on error. ASTERISK-24295 #close Reported by: Rogger Padilla Review: https://reviewboard.asterisk.org/r/3954/ ........ Merged revisions 423866 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	20f4ea0df7	chan_sip: Unref outbound proxy structure on dialog/pvt destruction. ... Make sure outbound proxy refs are always unreffed on dialog destruction. Review: https://reviewboard.asterisk.org/r/4016/ ........ Merged revisions 423800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423801 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423802 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	d25390073b	Make CDR and CEL unit tests less FRACKy. ... Prior to this commit, CDR and CEL tests were expected to trigger FRACKs (i.e. assertions) due to the fact that the channels they create have no formats on them. Some code was independently added recently that attempts to prevent FRACKs from occurring by failing early when attempting to set up translation paths if one or both channels support no formats. Unfortunately, this attempt to be helpful made the CDR and CEL tests go from simply FRACKing to outright failing and in some cases, failing so badly as to crash Asterisk. This commit seeks to correct past mistakes by adding the ulaw format to channels created by the CDR and CEL unit tests. This makes setting up translation paths succeed, eliminates previously-seen FRACKs, and ultimately causes the unit tests to succeed again. Review: https://reviewboard.asterisk.org/r/4014 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	0f3540553d	chan_sip: On INVITE retransmission, don't add an extra 503 response. ... INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is retransmitted, asterisk would generate a 503 in addition to the 486. Thanks Torrey Searle for providing a working regression test. ASTERISK-24335 #close Review: https://reviewboard.asterisk.org/r/4003/ Patches: retrans_486_invite.patch uploaded by Torrey Searle (License #5334) ........ Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423721 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423722 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9d1c0348f2	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. ... r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d85e59a23b	main/channel: Unlock channel in off-nominal path ... In r423414 (13) / r423415 (trunk), an API call that determines if a format capability structure is empty was added. This returns true if the format capability structure is completely empty or "none". A check for this was added in channel.c's set_format call. Unfortunately, when this check was true, it returned from the function while still holding the channel lock. This caused the CDR unit tests - which have a tendency to create channels with no formats - to deadlock. Whoops. This patch unlocks the channel on the off-nominal path. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f48d3f849d	rest-api/api-docs/events.json: Remove non-compliant 'extends' attribute ... Prior to the release of Swagger 1.2, the attribute 'extends' was being promoted as a possible way to show that a particular object extends an existing object. Instead, the Swagger specification went with the 'subTypes' attribute in the base object. This patch removes the unsupported attribute; the object that the offending objects proposed to extend already lists them in its 'subTypes' attribute. ASTERISK-24300 #close Reported by: Bradley Watkins ........ Merged revisions 423620 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d74dee638c	rest-api/api-docs: Correct basePath in resources to match top resources file ... The resources.json file that defines the resource JSON files used with ARI references a basePath of 'http://localhost:8088/ari'. This does not match what is defined in the resource files themselves, 'http://localhost:8088/stasis'. The correct base path is the one that includes 'ari' in the URL; this patch updates the various resource JSON files to have the correct basePath. ASTERISK-24339 #close Reported by: Bradley Watkins ........ Merged revisions 423617 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	9a988639f4	res_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist on reload. ... When unloading the module did not unregister the CLI commands causing a crash upon load when they were registered again. When reloading the module the return value from the config options framework was not checked to determine if an error occurred or not. This caused a message to be output saying the module did not exist when reloading if no changes were present. AST-1433 #close AST-1434 #close ........ Merged revisions 423579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fbbe455b9d	res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated. ... Outgoing PJSIP calls can result in non-negotiated formats listed in the channel's native formats if video formats are listed in the endpoint's configuration. The resulting call could then use a non-negotiated format resulting in one way audio. * Simplified the update of session->req_caps in set_caps(). Why do something in five steps when only one is needed? AFS-162 #close Review: https://reviewboard.asterisk.org/r/4000/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	d1b1e911bf	Stasis_channels: Resolve unfinished Dials when doing masquerades ... Masquerades into channels that are in the dialing state don't end their dial and this goes against the model for things like CDRs and generating Dial end manager actions and such. ASTERISK-24237 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c95b53e21a	chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings ... Caused by format changes in Asterisk 13 ASTERISK-24265 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/3999/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	fade256307	PJSIP: Prevent T38 framehook being put on wrong channel ... This change gives framehooks a reverse-direction masquerade callback in addition to chan_fixup_cb similar to the callback added to datastores to handle the same situation. The new callback provides the same parameters as the fixup callback, but is called on the new channel's framehooks before moving framehooks from the old channel to the new channel. This gives the framehooks an oppurtunity to decide whether they should remain on the new channel or be removed. This new callback is used to prevent the PJSIP T.38 framehook from remaining on a masqueraded channel if the new channel is not also a PJSIP channel. This was causing a crash when a local channel was masqueraded into a PJSIP channel and the framehook was executed on the local channel since the channel's tech private data was not structured as expected. Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	6b3c47bd6a	res_pjsip: Don't require a password when doing userpass authentication. ... An empty password is valid for username/password authentication so we should allow password to be empty/not supplied. Review: https://reviewboard.asterisk.org/r/3988 ........ Merged revisions 423481 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7ae706b2d	utils: Create ast_strsep function that ignores separators inside quotes ... This function acts like strsep with three exceptions... * The separator is a single character instead of a string. * Separators inside quotes are treated literally instead of like separators. * You can elect to have leading and trailing whitespace and quotes stripped from the result and have '\' sequences unescaped. Like strsep, ast_strsep maintains no internal state and you can call it recursively using different separators on the same storage. Also like strsep, for consistent results, consecutive separators are not collapsed so you may get an empty string as a valid result. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	23f58d6f80	Add subscription state test events. ... These are needed for a set of batched notification RLS tests that are about to be committed to the testsuite. Review: https://reviewboard.asterisk.org/r/3967 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5567d3e7d2	res_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDR ... Also fixes comma separates match lists ASTERISK-24290 #close Reported by: Ray Crumrine Review: https://reviewboard.asterisk.org/r/3995/ ........ Merged revisions 423417 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	588a8d9078	bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent. ... * Clarified some read/write format comments. * Fixed a doxygen tag typo. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8b0352ffae	astobj2.c/refcounter.py: Fix to deal with invalid object refs. ... * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	23a375be5f	Add API call to determine if format capability structure is "empty". ... Empty here means that there are no formats in the format_cap structure or the only format in it is the "none" format. I've added calls to check the emptiness of a format_cap in a few places in order to short-circuit operations that would otherwise be pointless as well as to prevent some assertions from being triggered in cases where channels with no formats are used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	5c5aee4458	res_fax_spandsp: Properly handle cleanup before starting FAXes. ... If faxing fails at a very early stage, then it is possible for us to pass a NULL t30 state pointer to spandsp, which spandsp is none too pleased with. This patch ensures that we pass the correct pointer to spandsp in the situation where we have not yet set our local t30 state pointer. ASTERISK-24301 #close Reported by Matt Jordan Patches: ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049) ........ Merged revisions 423360 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423365 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7a35800551	res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies. ... res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE arrives. * It checks that there is a subscription handler for the Event * It checks that there are body generators for the types in the Accept header The problem is, there's nothing that ensures that these two things will actually mesh with each other. For instance, Asterisk will accept a subscription to MWI that accepts pidf+xml bodies. That doesn't make sense. With this commit, we add some type information to the mix. Subscription handlers state they generate data of type X, and body generators state that they consume data of type X. This way, Asterisk doesn't end up in some hilariously mismatched situation like the one in the previous paragraph. ASTERISK-24136 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3877 Review: https://reviewboard.asterisk.org/r/3878 ........ Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a2482acdce	res_pjsip: ami: Fix error in AMI output when an endpoint has no transport ... When no transport is associated to an endpoint, the AMI output for PJSIPShowEndpoint indicates an error instead of silently ignoring the missing transport. This patch causes the error to appear only if a transport was specified on the endpoint and the transport doesn't exist. It also fixes an issue with counting the objects that were actually found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3998/ ........ Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	27df9b73e2	Only install dahdi_span_config_hook if DAHDI is enabled ... This patch changes the install to only install the hook script if DAHDI is enabled. It also adds the script to the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the _MAKEOPTS variables and their comment. This allows installs which specify a --prefix to work normally, as long as they don't enable DAHDI. Review: https://reviewboard.asterisk.org/r/3972/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	0a2e6a1c7e	config: bug: Fix SEGV in ast_category_insert when matching category isn't found ... If you call ast_category_insert with a match category that doesn't exist, the list traverse runs out of 'next' categories and you get a SEGV. This patch adds check for the end-of-list condition and changes the signature to return an int for success/failure indication instead of a void. The only consumer of this function is manager and it was also changed to use the return value. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3993/ ........ Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423277 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423278 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c48b609fb3	res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. ... ........ Merged revisions 423253 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423254 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	85d7e44186	res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. ... Side note: I need a vacation. ........ Merged revisions 423210 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	93f7c8a434	res_rtp_asterisk: Fix building when pjproject is not used. ... ........ Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	79b702f308	Voicemail: get correct duration when copying file to vm ... Changes made during format improvements resulted in the recording to voicemail option 'm' of the MixMonitor app writing a zero length duration in the msgXXXX.txt file. This change introduces a new function ast_ratestream(), which provides the sample rate of the format associated with the stream, and updates the app_voicemail function for ast_app_copy_recording_to_vm to calculate the right duration. Review: https://reviewboard.asterisk.org/r/3996/ ASTERISK-24328 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	48d58da883	res_pjsip_session: Fix usage of wrong memory pool when creating local SDP. ... ........ Merged revisions 423172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	4098d87eef	res_rtp_asterisk: Fix a myriad of TURN client issues. ... 1. The number of file descriptors an ioqueue instance can handle is fixed, so we now spawn the required number to handle the load. 2. Our transport identifiers were exceeding the range supported by pjnath. 3. The TURN client did not set up client binding causing needless bandwidth usage. 4. The code no longer updates address information on each packet. 5. STUN traffic was getting looped back to Asterisk instead of going through the TURN server. 6. Synchronization now ensures things are completely setup or destroyed. 7. Logging now reflects the target the TURN server is sending to/receiving from on our behalf. ASTERISK-23577 #close Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ ........ Merged revisions 423150 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423151 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	7efcca8de1	contrib: Fix verifyi typo in alembic DB script ps_transport table. ... Reported by: Zogot (on IRC) Patches: tmp.diff uploaded by Zogot, cleaned up by me. ........ Merged revisions 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	5840c47223	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. ... Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2e3f45e8e6	Realtime: Fix a bug that caused realtime destroy command to crash ... Also has could affect with anything that goes through ast_destroy_realtime. If a CLI user used the command 'realtime destroy <family>' with only a single column/value pair, Asterisk would crash when trying to create a variable list from a NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged revisions 422984 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	0f2bd8d855	Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. ... ast_play_and_record_full() has a parameter called "acceptdtmf" that is a string of acceptable DTMF digits that may be pressed by a caller to end and accept the recording. ARI uses this function in order to perform recording, and it provides options for what is passed as acceptdtmf to ast_play_and_record_full(). By default, ARI passes an empty string, with the intention that no DTMF can be used to end the recording. The problem is that ast_play_and_record_full() attempts to be "helpful" by setting "#" as the acceptdtmf if an empty string or NULL pointer has been passed in. With ARI, this results in unexpected behavior occurring if you have attempted to intercept "#" yourself in order to perform some other manipulation of the live recording. This change removes the "helpful" behavior by no longer accepting "#" as a default acceptdtmf if none is specified by the caller of ast_play_and_record_full(). This makes the ARI scenario work as expected. The other callers of ast_play_and_record_full() are app_voicemail and app_minivm, and in both cases, they pass an explicit "#" to ast_play_and_record_full() as acceptdtmf, so they are unaffected by this change. ........ Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago