asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	8fe29356ac	Fix dialplan function NULL channel safety issues (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	90fa4e3c36	main/formats: Fix crash in ast_format_cmp during non-clean shutdown. * Backport ast_register_cleanup from Asterisk 12. * Use ast_register_cleanup for format_attr_shutdown. ast_register_cleanup was originally commited in r390122 by dlee. (closes issue ASTERISK-23103) Reported by: JoshE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	818f476893	say: Fix a bug where SayNumber in Polish tries to play incorrect sound. This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ Merged revisions 411243 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	6a4e040aa8	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	336ecce920	chan_sip: Fix incorrect use of timers If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) ........ Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	4644a07334	chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted. (closes issue ASTERISK-20841) Reported by: Kelly Goedert ........ Merged revisions 411021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5b0f3c7458	app_confbridge: Fix bug - users with startmuted set don't start muted (closes issue ASTERISK-23461) Reported by: Chico Manobela Review: https://reviewboard.asterisk.org/r/3373/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	7199c911d9	res_fax_spandsp: Use g711_free() when available. Per Johann Steinwendtner on the asterisk-dev mailing list: http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a noop. I opted not to remove the call to g711_release() since it is harmless and to call g711_free() if we have a sufficiently recent version of spandsp. (issue ASTERISK-20149) Reported by: Alexandr Gordeev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russ Meyerriecks	d2e4c2208e	!fixup: callerid: Logic error in checksum processing Fixes syntax error in previous commit :-( ........ Merged revisions 410748 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russ Meyerriecks	4a0a04e5ca	callerid: Logic error in checksum processing Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) ........ Merged revisions 410710 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	83fe45376c	manager: fix memory leak in manager_add_filter function (closes issue ASTERISK-23420) Reported by: Etienne Lessard Patches: manager_eventfilter_leak uploaded by Etienne Lessard (license 6394) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	59ca55e7b7	Remove an extra ast_cond_wait() that slipped through the patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	0edf689519	Prevent delayed astdb syncs. The syncing thread sleeps for a second before waiting to be told to attempt to sync again. If a signal were sent during this sleeping period, we would end up having to wait until the next sync signal occurred in order to sync up the astdb. This code rearrangement also ensures that any pending transactions will be synced prior to Asterisk shutting down. Patches: db_sync.patch by John Hardin (License #6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4a3c8065ab	app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up while MOH is playing. When MOH is playing to a user in a conference and the user is kicked or hangs up from the conference then the AMI MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	771a703366	AST-2014-001: Stack overflow in HTTP processing of Cookie headers. Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0c014422c1	AST-2014-002: chan_sip: Exit early on bad session timers request This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 410308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	099b165e7b	chan_sip: Fix deadlock of monlock between unload_module and do_monitor Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ ........ Merged revisions 410224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	69fb0d2585	chan_sip: Allow static realtime members to be qualified during module load. When a static realtime peer with qualify=yes is loaded, Asterisk will fail to send an OPTIONS request due to the lastms being equal to 0. This results in the peer being unable to receive calls from Asterisk because the status is permanently UNKNOWN. This patch allows an OPTIONS request to be sent during module load by ignoring the lastms value on startup only. Review: https://reviewboard.asterisk.org/r/3294/ (closes issue ASTERISK-17523) Reported by: Maciej Krajewski Tested by: wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112) ........ Merged revisions 410105 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	fa1d3b0941	moh: fix a refcount error with realtime MOH I observed a crash in res_musiconhold on an Asterisk 11 system using realtime MOH. Investigation of the backtrace showed a corrupt mohclass, implying that it got destroyed before the code expected it to. I went looking for reference counting errors that could have caused this crash and this patch this result. It contains 2 changes. 1) Remove a usless block of code that was impossible to reach. There was even a comment indicating that it was impossible to reach. The conditional includes "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if block with the opposite check "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no good reason to keep it around. 2) A similar block to #1 contained a reference counting error. It stores state->class in the local variable mohclass without increasing its reference count. The reference count on mohclass is decremented at the end of the function. This block of code probably very rarely runs, which would help explain why this system was working fine for many months before experiencing a crash. Review: https://reviewboard.asterisk.org/r/3282/ ........ Merged revisions 410043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	69192903af	res_fax_spandsp: Fix crash when passing ulaw/alaw data to spandsp When acting as a T.38 fax gateway, res_fax_spandsp would at times cause a crash in libspandsp. This would occur when, during fax tone detection, a ulaw/alaw frame would be passed to modem_connect_tones_rx. That particular routine expects the data to be in slin format. This patch looks at the frame type and, if the data is ulaw/alaw, converts the format to slin before passing it to modem_connect_tones_rx. Review: https://reviewboard.asterisk.org/r/3296 (closes issue ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal Rybarik patches: spandsp_g711decode.diff uploaded by Michal Rybarik (license 6578) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	a14a4d5194	config: Fix inverted test The test of the result of the stat() call was inverted such that its output was only used if the call failed. This inverts the test so that the output of stat() is used correctly. This was causing full reloads on unchanged files. (closes issue ASTERISK-23383) Reported by: David Woolley ........ Merged revisions 409916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	185257db13	Fix documentation for PRESENCE_STATE to properly illustrate how to create a presence hint. There was a missing comma. This was discovered by Dan Kaplan. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	647c6859db	Corrected cross-platform stat nanosecond code When nanosecond time resolution was added for identifying config file changes, it didn't cover all of the myriad of ways that one might obtain nanosecond time resolution off of struct stat. Rather than complicate the #if even further figuring out one system from the next, this patch directly tests for the three struct members I know about today, and #ifdef's accordingly. Review: https://reviewboard.asterisk.org/r/3273/ ........ Merged revisions 409833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	c4c22efdeb	Blocked revisions 409436 ........ buildsystem: Unbreak 'make -qp' on 1.8. r408083 caused trouble with make -qp. Backport r408193 to 1.8 as well. (closes issue ASTERISK-23382) Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	4947a0b91b	Fix references to 'keys' CLI commands in astgenkey ........ Merged revisions 409777 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	c332bb27ed	Correct RTP handling in chan_unistim and fix transfer process broken in previous fix: - Fixed too early RTP setup with phone, that cause no ringback tone on caller side - Handle call transfer cancel only in STATE_CALL case (related to ASTERISK-23073) (Reported by: Németh Tamás, niurkin sil) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	62a4018771	Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'	11 years ago
Moises Silva	7f1450e32c	Fix res/res_http_websocket.c build failure in 32bit due to incorrect print format for uint64_t git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Moises Silva	9792c46a09	Fix WebRTC over WSS not working Several fixes for the WebSockets implementation in res/res_http_websocket.c * Flush the websocket session FILE* as fwrite() may not actually guarantee sending the data to the network. If we do not flush, it seems that buffering on the SSL socket for outbound messages causes issues * Refactored ast_websocket_read to take into account that SSL file descriptors may be ready to read via fread() but poll() will not actually say so because the data was already read from the network buffers and is now in the libc buffers (closes issue ASTERISK-23099) (closes issue ASTERISK-21930) Review: https://reviewboard.asterisk.org/r/3248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	506bfece74	func_audiohookinheritance: Check If A Channel Was Specified This patch prevents a crash when using the function audiohookinheritance without setting the channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by: Joel Vandal Patches: asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3272/ ........ Merged revisions 409623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bac36c6b82	AO2: Add an assert for bad objects This adds an assert that will only be active if Asterisk is compiled with DO_CRASH and allows the testsuite to fail tests that would otherwise require log file parsing. ........ Merged revisions 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	7e083b9781	res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE ICE sessions will now be restarted if sessions are changed to use new sets of remote candidates. (closes issue ASTERISK-22911) Reported by: Vytis Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b5b1e27266	rtp_engine: Clean up after a failed remote bridge Upon failure of an INVITE transaction meant to initiate a remote native bridge, rtp_engine.c would not clean up non-reference-counted bridge instance pointers leaving a dangling pointer which was being used to perform a local native bridge after the other channel had hung up. This lead to dereferencing into freed memory and plenty of AO2 errors. This change allows the remote native bridge loop to clean up properly when the bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy Laine ........ Merged revisions 409521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	51d8abf542	Minor whitespace change to 'sip show peers' output. (closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom ........ Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6224e9521a	doxygen: Tweak the link back to ye olde Digium website ........ Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	a223679ae1	Makefile: replace -O6 with -O3 -O6 is not a legal option of gcc. Unofficially gcc considers it to be equivalent of -O3. clang chalks on it, though. This commit sets the default optimization flag to be -O3, like gcc actually considered it. Review: https://reviewboard.asterisk.org/r/3280/ ........ Merged revisions 409308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bcf5878c71	chan_sip: Add precautionary p->owner checks. * Add precautionary p->owner checks in sip_hangup(), get_refer_info(), get_also_info(), and interpret_t38_parameters(). * Simplify some tangled logic in get_refer_info(), get_also_info(), and add_rpid(). * Removed some dead code in handle_request_invite(). (closes issue ASTERISK-23323) Reported by: Walter Doekes Patches: issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified) ........ Merged revisions 409207 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7f2bd4ea18	app_queue: Fix documentation generation The documentation for QueueMemberPaused was causing documentation generation to fail because the documentation for that AMI event was in the wrong location. This moves that documentation the correct location and adds a missing parameter. (closes issue SWDAT-261) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	16ef371cb9	chan_sip: Fix crash in ast_channel_hangupcause_set(). * Fix crash in ast_channel_hangupcause_set() because p->owner not checked before calling. Regression introduced by the fix for ASTERISK-22621. (closes issue ASTERISK-23135) Reported by: OK (issue ASTERISK-23323) Reported by: Walter Doekes ........ Merged revisions 409156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c382714769	res_rtp_asterisk: correct build error from r409129 Accidentally placed a declaration below functional code (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	8a7ef07f94	res_rtp_asterisk: Fix checklist creating problems in ICE sessions Prior to this patch, local candidate lists including SRFLX would fail to start properly when building ICE candidate check lists. This patch fixes that problem by making sure that each SRFLX candidate is associated with the proper base address so that the check list can create matches properly. This patch was written by jcolp. The issue will be left open to await testing by the issue participants. (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	280807c109	Fix memory stomping bug in astman. This memset complained in dev mod on my Ubuntu box. The memset is both unnecessary and dangerous. At this point, m hasn't been initialized yet, so the memset will write off to whatever address happens to be on the stack at the time. ........ Merged revisions 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	78d560157a	res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load. Change minrate from 2400 to 4800 on config reload in response to changes from ASTERISK-22790 only. Any config with minrate of 2400 that would fail before r405693 will still fail. Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ ........ Merged revisions 409052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6087765a48	rtp_engine: fix crash during remote native bridging when calling get_codecs When two RTP channels are in a remote bridge, the remote bridging loop in rtp_engine will periodically check to see if the two channels can still be bridged. One of the many things it checks is whether or not the codecs have changed on the channel. If the codec has changed, it will break out of the loop to re-determine which type of bridge is appropriate. In order to perform this check, the ast_rtp_glue virtual table's get_codec callback is called for each channel. The callback implementations assume that the channel tech private is valid when called; as such, there has always been some code in place to check whether or not the channel pvt is NULL before calling. However, this check is insufficient. The channels are unlocked during the remote bridging loop. It is possible for a channel to get masqueraded between the check for the pvt being NULL and the actual call to get_codec. When this occurs, the callback is called with a ZOMBIE channel, which now has a NULL pvt. Crash. While this has always been possible in Asterisk 1.8, it is much more likely to occur in Asterisk 11 and later versions due to the timing changes that occur when getting the codec from a channel. Note that this is much more likely to be reproduced on slow, boggy hardware running Asterisk 11 - but fairly rarely otherwise. Also Note: This crash was also caught by the various SIP blind transfer tests, in addition to the bug report Alec filed. Review: https://reviewboard.asterisk.org/r/3247/ (closes issue ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis ........ Merged revisions 409001 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@409002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	8882542192	configs/voicemail.conf.sample - Make mailcmd sample text more explicit Made the wording a bit more explicit. Didn't really change the meaning. ........ Merged revisions 408876 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	70184544f9	ignore AST_CONTROL_PVT_CAUSE_CODE without any messages (closes issue ASTERISK-23336) Reported by: Alexander Semych git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5a7221decb	Remove extra defines of AST_PBX_MAX_STACK. * Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ ........ Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	e2dae45080	app_forkcdr: ForkCDR v option does not keep CDR variables for subsequent records When the 'v' option is specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is set only for the first CDR in the chain. So ForkCDR works fine with this option only once. After the second and further calls to ForkCDR, CDR variables get cleared on all CDRs besides the first one and moved to the newly forked CDR. It always sets the KEEP_VARS flag on the first CDR in the chain, instead of the most recent CDR which is used as a base to fork a new CDR. This patch sets KEEP_VARS flag on the most recent CDR on the stack (the CDR used for forking). (closes issue ASTERISK-23260) Reported by: zvision Patches: app_forkcdr.diff uploaded by zvision (license 5755) ........ Merged revisions 408747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	380516fe2c	install_prereq: Missing uuid[-dev] for debian distros Added uuid and uuid-dev to install prereq script. (closes issue ASTERISK-23255) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	574fefa004	rtp_engine: Dynamic payload change in rtp mapping not supported Asterisk didn't support the dynamic payload change in rtp mapping in the 200 OK response. Scenario: Asterisk sends the INVITE proposing alaw and telephone-event, it proposes rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with alaw and telephone-event also, but it proposes a different rtpmap number (rtpmap:103) for telephone-event. Expected Behaviour: Asterisk should honour the rtpmapping in the response and send DTMF packets using 103 as payload type for DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload type 101. With this patch asterisk now supports changes that can occur in the rtp mapping in the response. (closes issue ASTERISK-23279) Reported by: NITESH BANSAL Review: https://reviewboard.asterisk.org/r/3225/ Patches: dynamic_payload_change.patch uploaded by nbansal (license 6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@408729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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