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${ noResults }
16832 Commits (d24e3e05bc6ba360d50f7af64652c01ee03e57a7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
d24e3e05bc |
Merged revisions 180007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6767bd053d |
Blocked revisions 179972 via svnmerge
........ r179972 | dvossel | 2009-03-03 16:01:24 -0600 (Tue, 03 Mar 2009) | 13 lines app_meetme not setting filename and fileformat correctly for realtime When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3cb51f0127 |
Fix a memory leak when updating a realtime member field.
This was discovered while looking at issue #14353 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
408082d376 |
Blocked revisions 179937 via svnmerge
........ r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179903 via svnmerge
........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179745 via svnmerge
........ r179745 | mmichelson | 2009-03-03 11:03:47 -0600 (Tue, 03 Mar 2009) | 8 lines Convert pbx_spool to use string fields instead of statically-sized buffers. In tests run after making this conversion, I noticed an approximate 85% reduction in memory usage for call file processing. Review: http://reviewboard.digium.com/r/168/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179609 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 151464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179465 via svnmerge
........ r179465 | russell | 2009-03-02 17:06:16 -0600 (Mon, 02 Mar 2009) | 4 lines Fix a reference leak in timerfd_set_rate(). (found during a debugging session with dvossel and mmichelson.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179402 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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KeepAlive application no longer exists, so fix gosub implementation to not use it.
(closes issue #14571) Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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If cdr registration somehow succeeds without a config file, don't crash.
(closes issue #14563) Reported by: alerios git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179323 via svnmerge
........ r179323 | file | 2009-03-02 10:28:09 -0400 (Mon, 02 Mar 2009) | 5 lines Do not try to remove a registration scheduled item if the scheduler context has already been destroyed. (closes issue #14580) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179291 via svnmerge
........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179254 via svnmerge
........ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Add error checking when updating the "paused" field of a realtime queue member.
This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to this commit. (closes issue #14338) Reported by: fiddur Patches: 14338.patch uploaded by mmichelson (license 60) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179164 via svnmerge
........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines Mark res_ais as experimental, as the binary event format is subject to change. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179154 via svnmerge
........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179122 via svnmerge
........ r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines Add reload support to chan_skinny. Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179123 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 179057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 179021 via svnmerge
........ r179021 | russell | 2009-02-27 09:51:56 -0600 (Fri, 27 Feb 2009) | 7 lines Fix downloading SIREN7 and SIREN14 sound packages. In passing, also fix downloading SLIN16 extra sound packages. (closes issue #14565) Reported by: jtodd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178871 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 178870 via svnmerge
........ r178870 | murf | 2009-02-26 10:45:22 -0700 (Thu, 26 Feb 2009) | 1 line These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 178801 via svnmerge
........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178768 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 178764 via svnmerge
........ r178764 | file | 2009-02-26 11:40:10 -0400 (Thu, 26 Feb 2009) | 5 lines Ensure there is a valid tone part before trying to play tones. (closes issue #14558) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 178446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Change include order to make compile on Centos 5 with DAHDI
If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the __s32 type doesn't get defined git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 178300 via svnmerge
........ r178300 | dvossel | 2009-02-24 11:42:37 -0600 (Tue, 24 Feb 2009) | 14 lines Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well. Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not. (closes issue #14427) Reported by: snuffy Patches: iax_show_trunks.diff uploaded by snuffy (license 35) 2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 178030 via svnmerge
........ r178030 | dvossel | 2009-02-23 11:59:55 -0600 (Mon, 23 Feb 2009) | 7 lines Changes the way keyrotation is enabled by default Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. (closes issue #14523) Reported by: mvanbaak ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 178061 via svnmerge
........ r178061 | mvanbaak | 2009-02-23 19:23:38 +0100 (Mon, 23 Feb 2009) | 3 lines update the new manager commands in chan_skinny to match chan_sip's headers. requested by oej. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |