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${ noResults }
16911 Commits (84ab8b996382be59faec0c108806a31b45cdc4db)
Author | SHA1 | Message | Date |
---|---|---|---|
|
84ab8b9963 |
Merged revisions 183321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
bce77e2674 |
Merged revisions 183244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
70eaf71dfe |
Merged revisions 183242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9628d62ffa |
Merged revisions 183172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 183028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 182521 via svnmerge
........ r182521 | seanbright | 2009-03-17 10:24:53 -0400 (Tue, 17 Mar 2009) | 3 lines Don't include a space before the optional extra text that may follow a help string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 182408 via svnmerge
........ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 182362 via svnmerge
........ r182362 | russell | 2009-03-16 15:53:21 -0500 (Mon, 16 Mar 2009) | 2 lines Update UPGRADE.txt and CHANGES for 1.6.3 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 182355 via svnmerge
........ r182355 | russell | 2009-03-16 15:35:58 -0500 (Mon, 16 Mar 2009) | 29 lines Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182172 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 182071 via svnmerge
........ r182071 | mmichelson | 2009-03-13 12:49:01 -0500 (Fri, 13 Mar 2009) | 3 lines Remove ast_ prefix from functions which are not public. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 181985 via svnmerge
........ r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line improve a bit of suboptimal code ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 181846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 181769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 181731 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 181665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 181661 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ef6ff7199a |
Merged revisions 181656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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41f3b9eb4a |
Merged revisions 181612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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13ba3f9128 |
Merged revisions 181542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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72353ade99 |
Merged revisions 181499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ed7ecdba6e |
Blocked revisions 181465 via svnmerge
........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9afc536f86 |
Merged revisions 181444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cefe73c073 |
Merged revisions 181428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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91eb9b614b |
Merged revisions 181424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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000375fcce |
Merged revisions 181371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181372 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ed9ddfd8a5 |
Merged revisions 181345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181352 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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be3fc819ab |
Merged revisions 181296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 181292 via svnmerge
........ r181292 | russell | 2009-03-11 11:19:38 -0500 (Wed, 11 Mar 2009) | 2 lines Replace contents of this doc with a pointer to its new home ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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95c0766715 |
add missing header file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 181244 via svnmerge
........ r181244 | mmichelson | 2009-03-11 09:28:40 -0500 (Wed, 11 Mar 2009) | 11 lines Fix segfault when dialing a typo'd queue If trying to dial a non-existent queue, there would be a segfault when attempting to access q->weight, even though q was NULL. This problem was introduced during the queue-reset merge and thus only affects trunk. (closes issue #14643) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cbaceb5473 |
Blocked revisions 181210 via svnmerge
........ r181210 | file | 2009-03-11 10:44:42 -0300 (Wed, 11 Mar 2009) | 3 lines Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet option is enabled. (reported by Vadim Lebedev on the asterisk-dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b9ae2618b6 |
Fix merge oops from 181137
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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368b57494b |
Merged revisions 181135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Blocked revisions 181099 via svnmerge
........ r181099 | russell | 2009-03-10 21:25:24 -0500 (Tue, 10 Mar 2009) | 2 lines tabs to spaces ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |