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git.ipfire.org Git - thirdparty/asterisk.git/log
Kevin P. Fleming [Tue, 18 Aug 2009 20:42:51 +0000 (20:42 +0000)]
Convert all release tags to Opsound music-on-hold.
For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@212958
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Kevin P. Fleming [Tue, 12 Dec 2006 23:20:55 +0000 (23:20 +0000)]
use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48431
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Kevin P. Fleming [Tue, 12 Dec 2006 23:20:48 +0000 (23:20 +0000)]
importing files for 1.4.0-beta4 release
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48430
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Kevin P. Fleming [Tue, 12 Dec 2006 23:20:13 +0000 (23:20 +0000)]
Creating tag for the release of asterisk-1.4.0-beta4
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48429
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Kevin P. Fleming [Tue, 12 Dec 2006 23:18:14 +0000 (23:18 +0000)]
when making a release, we can always use wget and we can't run the configure script to find that out...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48427
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Kevin P. Fleming [Tue, 12 Dec 2006 23:14:24 +0000 (23:14 +0000)]
use autotagged externals
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48426
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Kevin P. Fleming [Tue, 12 Dec 2006 23:14:17 +0000 (23:14 +0000)]
importing files for 1.4.0-beta4 release
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48425
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Kevin P. Fleming [Tue, 12 Dec 2006 23:03:40 +0000 (23:03 +0000)]
Creating tag for the release of asterisk-1.4.0-beta4
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.0-beta4@48424
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Kevin P. Fleming [Tue, 12 Dec 2006 05:18:11 +0000 (05:18 +0000)]
block addition of silence files... not needed here
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48404
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Joshua Colp [Tue, 12 Dec 2006 04:13:48 +0000 (04:13 +0000)]
Use S_OR in my previous app_voicemail. This is the way it should have been done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48401
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Matt O'Gorman [Mon, 11 Dec 2006 23:02:10 +0000 (23:02 +0000)]
new sounds package with 100% more silence
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48399
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Matt O'Gorman [Mon, 11 Dec 2006 22:11:35 +0000 (22:11 +0000)]
Merged revisions 48394 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2
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r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) | 4 lines
app_externalivr needs a real silence file, and additional
changes to add silence files into core instead of extra
patch provided by bug 8177 with minor additions.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48396
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Joshua Colp [Mon, 11 Dec 2006 21:31:23 +0000 (21:31 +0000)]
Return non-existant callerid handling to that which it was before. In 1.4 and trunk callerid can be allocated but not have any contents so we have to use ast_strlen_zero before passing it to the relevant functions. (issue #8567 reported by pabelanger)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48391
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Tilghman Lesher [Mon, 11 Dec 2006 05:37:09 +0000 (05:37 +0000)]
STRFTIME() does not actually require an argument (issue 8540)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48382
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Joshua Colp [Mon, 11 Dec 2006 05:36:45 +0000 (05:36 +0000)]
Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48381
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Joshua Colp [Mon, 11 Dec 2006 05:30:01 +0000 (05:30 +0000)]
Use the correct API call to say a device state changed. (Yes, I'm a nub.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48379
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Joshua Colp [Mon, 11 Dec 2006 04:57:38 +0000 (04:57 +0000)]
Don't access the conference structure after it has been freed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48377
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Tilghman Lesher [Mon, 11 Dec 2006 00:47:21 +0000 (00:47 +0000)]
Merged revisions 48374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines
When doing a fork() and exec(), two problems existed (Issue 8086):
1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(), causing Asterisk
signal handlers within the child to execute, which caused nasty race conditions.
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Steve Murphy [Sun, 10 Dec 2006 03:04:18 +0000 (03:04 +0000)]
Merged revisions 48371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line
This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels).
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Russell Bryant [Sat, 9 Dec 2006 15:59:42 +0000 (15:59 +0000)]
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48363
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Russell Bryant [Sat, 9 Dec 2006 15:47:01 +0000 (15:47 +0000)]
Blocked revisions 48361 via svnmerge
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r48361 | russell | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48362
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Russell Bryant [Thu, 7 Dec 2006 18:17:28 +0000 (18:17 +0000)]
Merged revisions 48356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 Dec 2006) | 3 lines
Ensure that the file position is not incremented beyond the total number of
files available for playback. (issue #8539, ulogic)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48357
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Steve Murphy [Thu, 7 Dec 2006 15:33:32 +0000 (15:33 +0000)]
Here lies the fixes that killed bug 8423 -- OriginateSuccess and OriginateError incomplete channel name. May it rest in peace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48349
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Olle Johansson [Wed, 6 Dec 2006 16:25:06 +0000 (16:25 +0000)]
Issue #8258 - fix handling of 487 being retransmitted to Asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48326
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Russell Bryant [Wed, 6 Dec 2006 16:15:45 +0000 (16:15 +0000)]
Merged revisions 48322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines
Fix the name of the rtignoreregexpire option in the sample configuration file.
(issue #8526, arkadia)
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Olle Johansson [Wed, 6 Dec 2006 12:27:43 +0000 (12:27 +0000)]
Don't send Contact on MESSAGE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48317
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Olle Johansson [Wed, 6 Dec 2006 12:17:31 +0000 (12:17 +0000)]
Block patch from 1.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48316
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Joshua Colp [Tue, 5 Dec 2006 20:45:28 +0000 (20:45 +0000)]
Regenerate configure from Qwell's last commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48281
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Jason Parker [Tue, 5 Dec 2006 20:42:52 +0000 (20:42 +0000)]
Fix curl version number testing to be much more friendly to non-bash shells.
Issue 8508, patch by me. This *SHOULD* be POSIX compliant now..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48279
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Olle Johansson [Tue, 5 Dec 2006 17:29:43 +0000 (17:29 +0000)]
Merging the invitestate-1.4 branch after successful testing.
Will check if I can solve this with less changes in 1.2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48270
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Olle Johansson [Tue, 5 Dec 2006 15:59:05 +0000 (15:59 +0000)]
Add missing s from another repository. (thanks jcmoore!)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48268
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Olle Johansson [Tue, 5 Dec 2006 12:39:30 +0000 (12:39 +0000)]
Updating sip.conf.sample with information about T38 not working
when chan_local or chan_agent is involved in the call.
I don't know how big a fix that would be to solve, but this is
the current state of affairs.
(Chan_sip currently checks if the other side of the bridge
has a SIP tech. We could/should implement another check,
possibly for udptl_write or some flag in the ast_channel
structure).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48264
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Tilghman Lesher [Tue, 5 Dec 2006 01:41:02 +0000 (01:41 +0000)]
Oops, forgot to release the odbc handle
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48254
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Tilghman Lesher [Tue, 5 Dec 2006 01:34:34 +0000 (01:34 +0000)]
Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines
If the recording in the database is too large, it will fail to retrieve with
an mmap error. Not too sure why this doesn't happen when we put it in the
database, also, but since that doesn't seem to be broken, I'm not going to fix
it (at least until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385)
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Jason Parker [Mon, 4 Dec 2006 21:48:41 +0000 (21:48 +0000)]
Fix an issue which didn't allow unavail/greet/busy/etc messages from being saved into ODBC (and probably IMAP).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48248
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Jason Parker [Mon, 4 Dec 2006 21:23:11 +0000 (21:23 +0000)]
Blocked revisions 48246 via svnmerge
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r48246 | qwell | 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines
Revert change from 8016 - this breaks other stuff... Needs further review.
Tip: When you've reported a bug about something and somebody has put up a
patch for it.. It's not a good idea to open a completely new bug and say that
something is broken because of the patch in the other bug - PLEASE mention
something in the bug where the patch was actually created.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48247
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Jason Parker [Mon, 4 Dec 2006 19:08:18 +0000 (19:08 +0000)]
Blocked revisions 48236 via svnmerge
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r48236 | qwell | 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines
Fix an issue where a message isn't saved correctly when using ODBC storage and reviewing a message.
Issue 8016 - patch by sokhapkin.
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Joshua Colp [Mon, 4 Dec 2006 18:16:31 +0000 (18:16 +0000)]
Blocked revisions 48233 via svnmerge
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r48233 | file | 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines
If the generic bridge tells us not to retry, and we have a frame to spit out then break the bridge. Props to markit in #asterisk-bugs for bringing this up.
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Jason Parker [Mon, 4 Dec 2006 17:54:46 +0000 (17:54 +0000)]
Add documentation to voicemail.conf.sample for ODBC storage.
Issue 8499 - patch by blitzrage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48230
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Jason Parker [Mon, 4 Dec 2006 17:43:24 +0000 (17:43 +0000)]
Attempt to document some of the dependencies that are needed for net-snmp
Issue 8499 - initial patch by blitzrage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48228
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Russell Bryant [Sun, 3 Dec 2006 06:34:14 +0000 (06:34 +0000)]
When "fetch" is in use, instead of "wget", --continue is not a valid option.
(issue #8451)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48223
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Olle Johansson [Sat, 2 Dec 2006 21:45:58 +0000 (21:45 +0000)]
- Removing one of two pieces of code to handle 481 response on INVITE
- Move handling of REFER response to handle_response_refer()
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48219
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Olle Johansson [Sat, 2 Dec 2006 11:32:51 +0000 (11:32 +0000)]
- Disable RTP hold timers while T.38 fax transmission happens
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
something that video phones support in the RTP stream.
I now this is a big architectual change at this stage for 1.4, but decided it was needed
to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample
Issue 7679 in the bug tracker. Please test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199
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Russell Bryant [Sat, 2 Dec 2006 03:50:58 +0000 (03:50 +0000)]
Backport the comment containing the warning regarding the limitations on the
usage of this function. It is thread safe, but not technically reentrant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48195
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Kevin P. Fleming [Fri, 1 Dec 2006 23:37:28 +0000 (23:37 +0000)]
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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Russell Bryant [Fri, 1 Dec 2006 23:16:28 +0000 (23:16 +0000)]
FreeBSD 6.1 does not include wget by default. However, it has fetch which will
work just fine for our purposes of downloading the sounds packages. So, check
for both wget and fetch and the configure script and use what was found to
download them. If neither one was found, and sound packages are selected that
must be downloaded, the install process will print out an informative error
message indicating the situation.
Also, fix a couple places where "make" was hard coded into some output messages
by replacing them with the $(MAKE) variable.
(issue #8451, initial patch by pabelanger, with additional modifications by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48190
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Jason Parker [Fri, 1 Dec 2006 20:25:51 +0000 (20:25 +0000)]
Merged revisions 48183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
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Jason Parker [Fri, 1 Dec 2006 20:24:32 +0000 (20:24 +0000)]
erm, hmm...I typod a merge...that went...strangely
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Jason Parker [Fri, 1 Dec 2006 20:19:44 +0000 (20:19 +0000)]
Merged revisions 48182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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Tilghman Lesher [Fri, 1 Dec 2006 19:38:59 +0000 (19:38 +0000)]
Double-unlock error (reported by blitzrage on IRC)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48179
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Olle Johansson [Fri, 1 Dec 2006 17:41:56 +0000 (17:41 +0000)]
- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states
- Remove support for T.38 early media, since it's impossible.
(Two patches in one - extra friday evening offer due to being off line from svn today... :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177
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Joshua Colp [Thu, 30 Nov 2006 21:18:24 +0000 (21:18 +0000)]
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48168
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Olle Johansson [Thu, 30 Nov 2006 20:51:04 +0000 (20:51 +0000)]
Issue 8319 - change noncecount before using it.
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Joshua Colp [Thu, 30 Nov 2006 20:28:19 +0000 (20:28 +0000)]
Blocked revisions 48161 via svnmerge
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r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines
Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky)
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Joshua Colp [Thu, 30 Nov 2006 20:07:55 +0000 (20:07 +0000)]
Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines
Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)
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Joshua Colp [Thu, 30 Nov 2006 19:05:14 +0000 (19:05 +0000)]
Merged revisions 48154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155)
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Joshua Colp [Thu, 30 Nov 2006 18:47:40 +0000 (18:47 +0000)]
Merged revisions 48151 via svnmerge from
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r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines
Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader)
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Joshua Colp [Thu, 30 Nov 2006 18:19:55 +0000 (18:19 +0000)]
Blocked revisions 48146 via svnmerge
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r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines
Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia)
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Joshua Colp [Thu, 30 Nov 2006 17:57:35 +0000 (17:57 +0000)]
Merged revisions 48142 via svnmerge from
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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
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Olle Johansson [Thu, 30 Nov 2006 14:29:39 +0000 (14:29 +0000)]
Explain status reports and
make codefreeze more happy :-)
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Olle Johansson [Thu, 30 Nov 2006 09:12:30 +0000 (09:12 +0000)]
Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line
in the REGISTER request. Imported from 1.2.
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Joshua Colp [Wed, 29 Nov 2006 21:05:17 +0000 (21:05 +0000)]
Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey)
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Olle Johansson [Wed, 29 Nov 2006 19:56:56 +0000 (19:56 +0000)]
Explain the use device status system implemented in SIP for subscriptions,
queues and manager a bit better.
Like in 1.2, you will get more detailed information if you set a call
limit for a device. When the call limit is reached, the status system will
report a device as busy.
For queues, setting a call limit per SIP device is propably a requirement.
In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.
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Joshua Colp [Wed, 29 Nov 2006 16:50:33 +0000 (16:50 +0000)]
Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)
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Olle Johansson [Wed, 29 Nov 2006 08:03:36 +0000 (08:03 +0000)]
Clarify RTP timers. Sorry, grandma.
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Joshua Colp [Wed, 29 Nov 2006 04:26:53 +0000 (04:26 +0000)]
Don't crash if the mailstream was not created.
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Jason Parker [Tue, 28 Nov 2006 18:26:53 +0000 (18:26 +0000)]
Export several more variables in top level Makefile. Inspired by issue 8438.
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Joshua Colp [Tue, 28 Nov 2006 16:57:16 +0000 (16:57 +0000)]
Merged revisions 48087 via svnmerge from
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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
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Joshua Colp [Mon, 27 Nov 2006 18:06:50 +0000 (18:06 +0000)]
Merged revisions 48053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 lines
Use the proper function to get the new message count instead of always using the filesystem. (issue #8421 reported by slimey)
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Tilghman Lesher [Mon, 27 Nov 2006 17:20:37 +0000 (17:20 +0000)]
Merged revisions 48045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 Nov 2006) | 2 lines
Random MOH wasn't really random (bug 8381)
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Russell Bryant [Mon, 27 Nov 2006 17:17:40 +0000 (17:17 +0000)]
Remove a couple of unused variables (issue #8380, casper)
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Joshua Colp [Mon, 27 Nov 2006 15:32:19 +0000 (15:32 +0000)]
Merged revisions 48037 via svnmerge from
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r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 lines
Do not reference the freed outgoing structure in the debug message. (issue #8425 reported by arkadia)
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Olle Johansson [Mon, 27 Nov 2006 06:41:33 +0000 (06:41 +0000)]
Change logging message
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Steve Murphy [Sun, 26 Nov 2006 00:26:16 +0000 (00:26 +0000)]
might as well also document the raw values of the flag vars
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Steve Murphy [Sun, 26 Nov 2006 00:01:34 +0000 (00:01 +0000)]
A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it.
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Olle Johansson [Sat, 25 Nov 2006 09:28:28 +0000 (09:28 +0000)]
Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable
subscription support totally or per peer in sip.conf with allowsubscribe = yes | no
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Steve Murphy [Fri, 24 Nov 2006 17:17:07 +0000 (17:17 +0000)]
bug 8189 posted this fix for main/translate.c for PLC
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Christian Richter [Fri, 24 Nov 2006 15:46:13 +0000 (15:46 +0000)]
Merged revisions 47968 via svnmerge from
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r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line
fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm
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Olle Johansson [Thu, 23 Nov 2006 11:01:47 +0000 (11:01 +0000)]
Don't allocate unused variable.
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Joshua Colp [Wed, 22 Nov 2006 21:47:43 +0000 (21:47 +0000)]
Video will never reach Packet2Packet bridging and can do more harm then good.
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Steve Murphy [Wed, 22 Nov 2006 02:37:26 +0000 (02:37 +0000)]
blocking 47910 from 1.4; the time-limits on Dial are working OK
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Joshua Colp [Tue, 21 Nov 2006 17:32:27 +0000 (17:32 +0000)]
If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate)
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Olle Johansson [Tue, 21 Nov 2006 15:20:03 +0000 (15:20 +0000)]
Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as
early media. (discovered at the AVTF meeting in Paris).
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Tilghman Lesher [Mon, 20 Nov 2006 20:01:58 +0000 (20:01 +0000)]
Oops, merge missed release of odbc object
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Tilghman Lesher [Mon, 20 Nov 2006 20:00:54 +0000 (20:00 +0000)]
Merged revisions 47862 via svnmerge from
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r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) | 2 lines
Failing to trap -1 error from mmap causes segfault (Issue 8385)
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Joshua Colp [Mon, 20 Nov 2006 19:51:36 +0000 (19:51 +0000)]
Merged revisions 47859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines
Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs)
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Joshua Colp [Mon, 20 Nov 2006 16:17:47 +0000 (16:17 +0000)]
Blocked revisions 47855 via svnmerge
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r47855 | file | 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines
Free history items at the end of use of the temporary SIP pvt structure. (issue #8383 reported by benh)
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Joshua Colp [Mon, 20 Nov 2006 15:58:50 +0000 (15:58 +0000)]
Only remove/destroy the RTCP I/O item if it exists.
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Joshua Colp [Mon, 20 Nov 2006 15:51:37 +0000 (15:51 +0000)]
Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings)
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Olle Johansson [Mon, 20 Nov 2006 11:45:22 +0000 (11:45 +0000)]
Explain properly how videosupport works.
Committ from Asterisk Video Task Force meeting in Paris!
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Olle Johansson [Mon, 20 Nov 2006 11:29:19 +0000 (11:29 +0000)]
Make sure we destroy scheduled items and not use them ever again after destruction (rizzo)
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Luigi Rizzo [Sat, 18 Nov 2006 17:59:35 +0000 (17:59 +0000)]
fix bug 7450 - Parsing fails if From header contains angle brackets
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
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Kevin P. Fleming [Fri, 17 Nov 2006 19:02:38 +0000 (19:02 +0000)]
block patch that was already present here
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Jason Parker [Thu, 16 Nov 2006 23:19:46 +0000 (23:19 +0000)]
Fix a couple of typos. Initially pointed out by mrobinson.
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Jason Parker [Thu, 16 Nov 2006 23:18:22 +0000 (23:18 +0000)]
Blocked revisions 47780 via svnmerge
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r47780 | qwell | 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines
Fix a couple of typos in applications.. Initially spotted by mrobinson.
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Kevin P. Fleming [Thu, 16 Nov 2006 23:00:10 +0000 (23:00 +0000)]
update documentation regarding IAX2 transfers and CDRs
Merged revisions 47776 via svnmerge from
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r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) | 2 lines
update clearly wrong documentation regarding cdr_custom
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Joshua Colp [Thu, 16 Nov 2006 21:11:06 +0000 (21:11 +0000)]
Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath)
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Joshua Colp [Thu, 16 Nov 2006 20:30:54 +0000 (20:30 +0000)]
Blocked revisions 47761 via svnmerge
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r47761 | file | 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines
Look for the header file specifically in all cases, not just the existence of the directory. (issue #8358 reported by mrness)
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Kevin P. Fleming [Thu, 16 Nov 2006 20:09:10 +0000 (20:09 +0000)]
check for pre-1.4 versions of Zaptel and abort the configure script if found with an appropriate error message
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Olle Johansson [Thu, 16 Nov 2006 19:24:23 +0000 (19:24 +0000)]
Make the HOLD notification optional, in order to avoid a lot of extra database lookups
for all those realtime users out there.
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projects / thirdparty / asterisk.git / log
