Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
12000 Commits
150 Branches
1601 Tags
295 MiB
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from 'dd15e9dc7b'
${ noResults }
Commit Graph

12000 Commits (dd15e9dc7b924ae378d64f16a774ecdedb8d0d8c)
 

Author SHA1 Message Date
Olle Johansson fc49ddab3a Improve support for multipart messages. Code by gasparz, changes
18 years ago
Tilghman Lesher 1af09c5f9d When a recording ends with '#', we are improperly trimming an extra 200ms from the recording.
18 years ago
Joshua Colp f3f12761ba Return the proper value when the srv_callback function executes properly.
18 years ago
Jason Parker fb0bb38fc4 Fix building on newer systems which require a third arg to open() when using O_CREAT.
18 years ago
Jason Parker b0e9d400ff Revert change from revision 67064.
18 years ago
Tilghman Lesher 4332b72082 If we set a value for qualify, we should actually pay attention to it, instead of overriding the value
18 years ago
Mark Michelson dc6e3e9d5d Reverting commit made in revision 89205 since it is unnecessary.
18 years ago
Tilghman Lesher feed493993 Debugging is running into the 16-lock limit. Increase to avoid.
18 years ago
Mark Michelson 4155b5f984 Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options
18 years ago
Jason Parker 05df1092da Fix a typo pointed out by De_Mon on #asterisk-dev
18 years ago
Tilghman Lesher 472eb33648 If two config writes collide, file corruption could result. Use a mkstemp() file, instead.
18 years ago
Tilghman Lesher 1a052e0498 Fix two cases of memory corruption caused by background threads.
18 years ago
Christian Richter c2c1e68238 if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
18 years ago
Christian Richter 4e52dc67dc added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it.
18 years ago
Christian Richter 472f7a471c fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all)
18 years ago
Christian Richter ad50f139c4 fixed the support for CW and therefore for the reject_cause option.
18 years ago
Christian Richter 57ccb76df1 aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
18 years ago
Jason Parker 76aa7c3767 Properly say the seconds here..
18 years ago
Mark Michelson 3b75ff9010 Rework of the commit I made yesterday to use the already built-in
18 years ago
Jason Parker c170f694e7 Avoid warnings on load when using sample configuration files.
18 years ago
Mark Michelson 9f5cf47a6f I made this same adjustment in trunk to fix a bug, and it makes sense to do it in 1.4 as
18 years ago
Kevin P. Fleming 30cb593fd2 fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting
18 years ago
Tilghman Lesher 7e81a39a81 Typo
18 years ago
Joshua Colp 3aea241b63 Do not add a sip: to the beginning of the To URI unless needed.
18 years ago
Joshua Colp 0f1ef85f9a Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
18 years ago
Joshua Colp 53fd91490e Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support.
18 years ago
Joshua Colp f1309f2c3f If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan.
18 years ago
Tilghman Lesher 45c16cc29b The member refcount must be incremented, to avoid using it after deallocation.
18 years ago
Mark Michelson 0d76379f54 This patch makes it possible for SIP phones to dial extensions defined with '#' characters
18 years ago
Steve Murphy 164d8a5e61 In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
18 years ago
Mark Michelson 63f6e6b023 Fixing a segfault in the manager "core show channels concise" command.
18 years ago
Tilghman Lesher 645af85225 Suppress AEL warnings on load.
18 years ago
Russell Bryant 6e74f69b51 Fix init_classes() so that classes that actually do have files loaded aren't
18 years ago
Jason Parker a0edd3f3f3 Correctly set the total number of channels from a zaptel transcoder board.
18 years ago
Tilghman Lesher f75916e7be We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops).
18 years ago
Olle Johansson 044aa79799 Bug fixes to tdd support in zaptel.
18 years ago
Russell Bryant 3946288786 If someone were to delete the files used by an existing MOH class, and then
18 years ago
Steve Murphy 712b337863 closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix.
18 years ago
Joshua Colp cac21aa19b Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
18 years ago
Joshua Colp 8b309c7bb9 Fix improbable but possible memory leaks in chan_zap.
18 years ago
Russell Bryant c60344fd8e Remove some checks to see if locks are initialized from the non-DEBUG_THREADS
18 years ago
Kevin P. Fleming 2c76da2828 update comment to match the state of the code
18 years ago
Mark Michelson 7b59a194a5 Reworked deadlock avoidance in __ast_read. Restored audio to
18 years ago
Russell Bryant 34002d567b After seeing crashes related to channel variables, I went looking around at the
18 years ago
Russell Bryant ea00780d49 When traversing the list of channel variables here in transmit_invite(), the
18 years ago
Russell Bryant 69e42e6096 Fix up some indentation.
18 years ago
Russell Bryant 5d140cb9c2 Merge changes from asterisk/team/kpfleming/SRV-priority-handling
18 years ago
Russell Bryant 9cb94c7cde Merge the last bit of changes from asterisk/team/russell/readq-1.4
18 years ago
Joshua Colp 10c172eb00 If a SIP channel is put on hold multiple times do not keep incrementing the onHold value.
18 years ago
Russell Bryant a3af50b67d Fix up datastore handling in ast_do_masquerade(). The code is intended to move
18 years ago