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${ noResults }
2272 Commits (cf7e1de2d19eb941ccbe502da5bf1166e439490d)
Author | SHA1 | Message | Date |
---|---|---|---|
|
2c371ff2b7 |
Merged revisions 160481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
107dfe787e |
Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) ........ ................ r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b0bad74445 |
Merged revisions 152216,152287,152369,152467,152569,152605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) | 13 lines Merged revisions 152215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) ........ ................ r152287 | jpeeler | 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged revisions 152286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines Buffer policy setting for half is not needed. ........ ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy ........ ................ r152467 | tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 lines Merged revisions 152463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines Quoting in the wrong direction (Fixes AST-107) ........ ................ r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) ........ ................ r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | 22 lines Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 160308 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 159818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@159855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158265-158266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines Use some magic constants to get the right size for this sscanf statement. Thanks Richard! ........ r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines Use a more expressive constant for a 64-bit scanned int ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Once again, Russell to the rescue. Use the builtin astobj1 lock
of the sip_peer and sip_user instead of adding a new one git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 155467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@155468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fa635ea4b0 |
port gcc 4.3.x warning fixes from trunk to this branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4ebe1cfd46 |
Fix some memory leaks. These issues are 1.6.0 specific.
- Freeing the peer got accidentally removed from the peer's destructor. It is still needed for astobj, but not for astobj2. - Fix some places that called find_user or find_peer, but did not release the reference that was returned. (closes issue #13331) Reported by: sergee Patches: chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138) Tested by: sergee git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@151765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6e93187aa1 |
Instead of merging commit 150307 to 1.6.0, I had
meant to block it in 1.6.1...time to go home :) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@150313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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258c4d4932 |
Merged revisions 150307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 | mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 lines After a long discussion on #asterisk-bugs, it seems kind of odd that a channel would be named after the port on which it came in on. For endpoints that always include ":5060" as part of the From: header, it will mean that you have a ton of channels with names like "SIP/5060-3ea38a8b." I am boldly moving forward with this change in trunk, but I'm not touching other branches with this one since this definitely would qualify as a behavior change. If there is a problem with this commit, and I haven't seen the obvious reason why you'd want to name the channel after the port from which the call originated, then please feel free to revert this ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@150308 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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962a134bcf |
Merged revisions 149487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct 2008) | 9 lines Merged revisions 149452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@149495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 148373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines Make sure that the inUse and inRinging fields for a sip peer cannot go below zero. This is a regression from 1.4 and so it will be applied to 1.6.0 as well. (closes issue #13668) Reported by: mjc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5b1e55af1e |
Merged revisions 145249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r145249 | jpeeler | 2008-09-30 17:21:19 -0500 (Tue, 30 Sep 2008) | 6 lines (closes issue #13337) Reported by: pj Tested by: pj Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@145255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3919045ee6 |
Merged revisions 144025 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r144025 | mmichelson | 2008-09-23 10:37:00 -0500 (Tue, 23 Sep 2008) | 16 lines When a promiscuous redirect contained both a user and host portion in the Contact URI and specifies a transport, the parsing done in parse_moved_contact resulted in a malformed URI. This commit fixes the parsing so that a proper Dial string may be formed when the forwarded call is placed. (closes issue #13523) Reported by: mattdarnell Patches: 13523v2.patch uploaded by putnopvut (license 60) Tested by: mattdarnell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9f74603219 |
Merged revisions 143559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r143559 | murf | 2008-09-18 17:41:33 -0600 (Thu, 18 Sep 2008) | 9 lines Merged revisions 143534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 line A micro-fix, in sip_park_thread, where d is freed before the func is done using it. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@143560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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86598fd373 |
Merged revisions 142866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines Create rules for disallowing contacts at certain addresses, which may improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0b963bfc68 |
Merged revisions 141566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@141567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b2f6976dbc |
Revert commit 140302. Should not be merging
changes like that into a release-candidate branch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f94f437af6 |
Merged revisions 140301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines Merged revisions 140299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when in pedantic mode. The problem was that the wrong tags would be compared depending on the direction of the call. (closes issue #13353) Reported by: flefoll Patches: chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0de177bc47 |
Merged revisions 139870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) | 10 lines Merged revisions 139869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines Make SIPADDHEADER() propagate indefinitely ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f084b49eed |
Merged revisions 139016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 138778-138780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line While we're at it, make this machine parseable too. ........ r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we don't need anymore. ........ r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, too (woops) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138781 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 138775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 lines Change event header to RegistrationTime to be more consistent (and avoid breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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eec03bcf40 |
Merged revisions 138260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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17f520666d |
Merged revisions 137812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines Make sure we set the socket port, so we don't try to use <ip address>:0. (closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@137814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 137532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines Correctly end locally ended calls. (closes issue #12170) Reported by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36) Tested by: bbryant, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@137533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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45253fa663 |
Merged revisions 135128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines Picky, picky, buildbot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 135126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines SIP should use the transport type set in the Moved Temporarily for the next invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 134556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 133573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul 2008) | 15 lines Merged revisions 133572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines We need to make sure to null-terminate the "name" portion of SIP URI parameters so that there are no bogus comparisons. Thanks to bbryant for pointing this out. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 133568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines Minor coding guidelines tweaks ... - Use ast_strlen_zero in one place - check for successful string comparison the way most of Asterisk code does it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 133509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) | 11 lines Merged revisions 133488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines Fix rtautoclear and rtcachefriends (Closes issue #12707) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 133197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 132823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines Well, the content of a channel variable may be longer than the size of a pointer... Thanks, eliel! Reported by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) (closes issue #13135) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 132795 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ Allow Spiraled INVITEs to work correctly within Asterisk. Prior to this change, a spiraled INVITE would cause a 482 Loop Detected to be sent to the caller. With this change, if a potential loop is detected, the Request-URI is inspected to see if it has changed from what was originally received. If pedantic mode is on, then this inspection is fully RFC 3261 compliant. If pedantic mode is not on, then a string comparison is used to test the equality of the two R-URIs. This has been tested by using OpenSER to rewrite the R-URI and send the INVITE back to Asterisk. (closes issue #7403) Reported by: stephen_dredge Modified: branches/1.4/channels/chan_sip.c ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 132703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines Merged revisions 132645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines The most common question on the #asterisk iRC channel and on mailing lists seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 132468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing a tcpbindaddr if it was disabled. (closes issue #13117) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 131044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) | 16 lines Merged revisions 130959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 130145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk Merging this rev from trunk to 1.6.0 was not simple. Why? Because we've enhanced trunk to do a [fast] merge-and-delete operation which also solved problems with contexts having entries from different registrars. Fast as in the amount of time the contexts are locked down. That *is* fast, but traversing the entire dialplan looking for priorities to delete takes more time overall. This particular fix involved pulling in those enhancements from trunk, along with all the various fixes and refinements made along the way. Merging all this from trunk into 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert all but the prop changes c. catalog all revisions to pbx.c since 1.6.0 was forked (at rev 105596). d. catalog all revisions to pbx.c in trunk since 1.6.0 was forked, making special note of all revs that were not merged into 1.6.0. e. study each rev in trunk not applied to 1.6.0, and determine if it was involved in the merge_and_delete enhancements in trunk. 25 commits were done in 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 additional changes, of which 7 were involved in the merge_and_delete enhancements: 106757 108894 109169 116461 123358 130145 130297 f. Go to trunk and collect patches, one by one, of the changes made by each rev across the entire source tree, using svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0, and resolve all failures and compilation problems before proceding to the next patch. h. test the stuff. i. profit! ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested by: murf (closes issue #12960) Reported by: mnicholson In this 'omnibus' fix, I **think** I solved both the problem in 13041, where unloading pbx_ael.so caused crashes, or incomplete removal of previous registrar'ed entries. And I added code to completely remove all includes, switches, and ignorepats that had a matching registrar entry, which should appease 12960. I also added a lot of seemingly useless brackets around single statement if's, which helped debug so much that I'm leaving them there. I added a routine to check the correlation between the extension tree lists and the hashtab tables. It can be amazingly helpful when you have lots of dialplan stuff, and need to narrow down where a problem is occurring. It's ifdef'd out by default. I cleaned up the code around the new CIDmatch code. It was leaving hanging extens with bad ptrs, getting confused over which objects to remove, etc. I tightened up the code and changed the call to remove_exten in the merge_and_delete code. I added more conditions to check for empty context worthy of deletion. It's not empty if there are any includes, switches, or ignorepats present. If I've missed anything, please re-open this bug, and be prepared to supply example dialplan code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 130167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 line Missed one. Formatting only. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 130129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |