21861 Commits (9333071c1f6639095b033a797f3cf2b7a4fc445a)

 

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Author	SHA1	Message	Date
Matthew Jordan	9333071c1f	Fixed invalid memory access when adding extension to pattern match tree ... When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342770 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4b826c46b3	Don't crash on empty notify channel ... ........ Merged revisions 342715 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	85dbd68f87	Whitespace and some better macro variable names. ... * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to __extracted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f7ce570c35	Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. ... AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342662 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	849992fde9	tweak the v21 detector to detect an additional pattern of hits and misses ... ........ Merged revisions 342605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	e5ac65bb43	Fix sequence number overflow over 16 bits causing codec change in RTP packets. ... Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342603 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	b61256c64b	Cleanup reference leaks in res_jabber ... res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342546 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ebf860e157	Check fopen return value for ao2 reference debug output. ... Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ ........ Merged revisions 342487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342488 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6cf08ad550	Change D-channel warning to be less confusing on non-NFAS setups. ... The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342485 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6e730a6806	Use int for storing ao2_container_count instad of size_t ... AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342436 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	f8351a8342	Simplify queue membercount code ... Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342384 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5749ef5be8	Properly update membercount for reloaded members ... Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342381 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	3551bfcbbc	Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c ... One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. ........ Merged revisions 342328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342329 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	c34674511d	Merged revisions 342277 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r342277 | kmoore | 2011-10-25 11:08:04 -0500 (Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines Fix spool handling to allow call files to be hardlinked into place This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	700f0a2752	Return NULL when no results returned for realtime_multientry ... It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342224 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	41ade83749	Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines. ... ........ Merged revisions 342183 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f6cd5af36b	Fixes a segfault caused by referencing null frames introduced in r338623 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3e9f1ee3e0	Fix use of OBJ_KEY in Queue application. ... To use the new OBJ_KEY flag, the container hash and compare callback functions must be updated to support OBJ_KEY. Otherwise, bad things happen. (issue ASTERISK-14769) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	05c6628c55	Outbound SIP OPTIONS messages will now include fromuser of related peer. ... This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) ........ Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342062 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	7ac53e57b3	queues container needs locking when using the OBJ_NOLOCK flag ... ........ Merged revisions 342017 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	3d55a05019	Remove some ref leaks and a return without unlock. ... There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. ........ Merged revisions 341972 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	d36c70e021	Whitespace Fixups / Add Braces ... This janitorial patch is related to work on RB1538 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	97a78b6234	Merged revisions 341313 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed, 19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	f39cbc004d	only process args that exist ... ASTERISK-18395 ........ Merged revisions 341809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9c7a017540	don't limit the length of app and function arguments ... ASTERISK-18395 ........ Merged revisions 341806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341807 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	b009ea5216	White space fixes in res_fax ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b961d57c4c	Fix AGI exec Park to honor the Park application parameters. ... The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	0e887d1a50	Fixed typo from previous commit ... ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5f5e908b19	Updated documentation for the optional CID parameter with CALLERID ... ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341665 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	e2e6e511af	Merged revisions 341599 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20 Oct 2011) | 8 lines add documentation for check_state_unknown in configs/queues.conf.sample app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	71b7df16bf	Merged revisions 341580 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	2f1130e13f	Clean up ast_check_digits ... The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	3f98c937a1	Merged revisions 341486 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) | 18 lines Fix a performance regression introduced in r325483. The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	1ed8cd087a	Outgoing calls with Google Voice ... Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341436 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5f8648892f	Don't use is_int() since it doesn't link well on all platforms ... Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Stefan Schmidt	2816ccc516	Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ... ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b0076c5be1	Don't resolve numeric hosts or contact unresolved hosts ... If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	10de040b6e	More parking issues. ... * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	d19ddf8741	Remove an unused include of md5.h ... Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 Merged-From: http://svn.asterisk.org/svn/asterisk/branches/10@341148 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	9f83c2b513	Initialize variables before calling parse_uri ... If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	ca0f2acab7	Set 'core' support level for test_format_api.c ... ........ Merged revisions 341146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	2ffea6ddc3	Multiple revisions 341108,341112 ... ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341122 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	a79c41ee66	Add information about limitations of new codec support in channel drivers. ... (issue ASTERISK-18680) ........ Merged revisions 341094 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	2cb5178d29	Don't try to remove peers without IPs from peers_by_ip ... (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341089 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	c292e39cdc	Change the internal name of the menuselect options that are used to control ... whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341023 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	899df042f5	Fix simple switch to not progress a call when call already progressed. ... If a simple switch was started on a device and then a specific call made (such as redial or speed dial), on timeout of the simple switch the call would be attempted again. This patch only allows the simple switch to make a call if the substate is still in the collecting digits mode. Also added small debug message to dialAndAactivate sub. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	4b9546abdf	Merged revisions 340971 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	e77f1a6ae1	Some additional module documentation changes for 10 for the menuselect change. ... (issue ASTERISK-18268) ........ Merged revisions 340931 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	19d3e269f6	Avoid unnecessary WARNING message ... Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340879 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cabf08b511	Fix DTMF blind transfer continuing to execute dialplan after transfer. ... Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago