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git.ipfire.org Git - thirdparty/asterisk.git/log
Steve Murphy [Mon, 4 Jun 2007 21:59:34 +0000 (21:59 +0000)]
This typo has been here since 1.4 forked. It has been the source of heartburn to many a dialplan/CDR programmer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67073
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Russell Bryant [Mon, 4 Jun 2007 21:47:36 +0000 (21:47 +0000)]
Add a missing \n. (pointed out by jcmoore on IRC)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67071
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Joshua Colp [Mon, 4 Jun 2007 19:31:09 +0000 (19:31 +0000)]
Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67068
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Joshua Colp [Mon, 4 Jun 2007 17:59:14 +0000 (17:59 +0000)]
Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67066
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Joshua Colp [Mon, 4 Jun 2007 17:41:59 +0000 (17:41 +0000)]
Returning a value that indicates the parking of a call was a success when it really wasn't (because the parking slot selected was in use) is the wrong thing to do. (issue #9723 reported by mdu113)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67064
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Tilghman Lesher [Mon, 4 Jun 2007 17:11:43 +0000 (17:11 +0000)]
Merged revisions 67060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) | 2 lines
Add revision Id tags (by request of tzafrir)
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Russell Bryant [Mon, 4 Jun 2007 16:02:31 +0000 (16:02 +0000)]
Change the configure script to build a test program against libcurl to make
sure the results from curl-config can be used to compile successfully. This is
intended to help prevent a situation where you are cross compiling, and the
configure script finds the curl library installed on the host.
(issue #9865, reported and patched by zandbelt)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67026
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Tilghman Lesher [Mon, 4 Jun 2007 15:50:16 +0000 (15:50 +0000)]
Issue 9739 - Malformed jid causes a crash
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67021
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Russell Bryant [Mon, 4 Jun 2007 15:47:40 +0000 (15:47 +0000)]
Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that
was marked as the final transmission for a call, don't call iax2_destroy() for
that call while the global frame queue is still locked. There is a very nice
explanation of the deadlock in the report.
(issue #9663, thorough report and patch from stevedavies, additional positive
test reports from mihai and joff_oconnell)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67020
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Russell Bryant [Mon, 4 Jun 2007 15:28:33 +0000 (15:28 +0000)]
Fix some compiler warnings in C++ modules.
(issue #9866, reported by osk, patch by Corydon76)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67018
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Tilghman Lesher [Fri, 1 Jun 2007 21:45:44 +0000 (21:45 +0000)]
On some drivers, deallocating the statement handle isn't enough. We also have to clear the cursor (nice, Oracle)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66919
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Mark Michelson [Fri, 1 Jun 2007 21:31:49 +0000 (21:31 +0000)]
Removing extraneous debugging lines from revision 66897. Sorry :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66917
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Mark Michelson [Fri, 1 Jun 2007 21:09:30 +0000 (21:09 +0000)]
Submitting a fix for voicemail with IMAP storage. Attachments with format specified as gsm were duplicated (i.e. two attachments) were left.
Thank you very much to xmarksthespot for submitting the patch that fixed this. (Issues 9787 and 8873, Reported by xmarksthespot and jerjer, patched by xmarksthespot)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66897
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Russell Bryant [Fri, 1 Jun 2007 19:41:30 +0000 (19:41 +0000)]
Changes to the way DTMF is handled in the core broke dialing in chan_skinny.
This patch makes chan_skinny usable again. I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was
written by DEA)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66881
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Russell Bryant [Fri, 1 Jun 2007 19:35:13 +0000 (19:35 +0000)]
List app_meetme as a module that app_page depends on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66879
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Tilghman Lesher [Thu, 31 May 2007 23:03:28 +0000 (23:03 +0000)]
Issue 9850 - update preferred command line syntax
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66821
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Russell Bryant [Thu, 31 May 2007 18:41:58 +0000 (18:41 +0000)]
Change a couple of header files to not use "new", which is a reserved keyword
in C++. (issue #9830, reported by osk)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66775
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Tilghman Lesher [Thu, 31 May 2007 17:15:09 +0000 (17:15 +0000)]
Merged revisions 66744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) | 2 lines
Issue 9818 - Fix for issue 8329 breaks pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, but only because we lack core API to do it.
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Joshua Colp [Thu, 31 May 2007 16:14:48 +0000 (16:14 +0000)]
Merged revisions 66764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines
It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)
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Mark Michelson [Wed, 30 May 2007 23:26:39 +0000 (23:26 +0000)]
Fixed seg-faults when recording greetings in voicemail with IMAP enabled. (Issue No. 9735, reported by xmarksthespot, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66671
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Joshua Colp [Wed, 30 May 2007 17:28:12 +0000 (17:28 +0000)]
Silly me for having out of date source! Oh well... I'm still leaving my comment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66639
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Joshua Colp [Wed, 30 May 2007 17:21:06 +0000 (17:21 +0000)]
When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66637
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Joshua Colp [Wed, 30 May 2007 16:06:37 +0000 (16:06 +0000)]
Change how channel names are generated a bit. (issue #9825 reported by eldadran)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66602
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Tilghman Lesher [Tue, 29 May 2007 21:56:07 +0000 (21:56 +0000)]
Merged revisions 66537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) | 2 lines
If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1.
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Olle Johansson [Tue, 29 May 2007 19:32:57 +0000 (19:32 +0000)]
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66503
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Olle Johansson [Tue, 29 May 2007 19:02:04 +0000 (19:02 +0000)]
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66474
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Joshua Colp [Tue, 29 May 2007 16:44:34 +0000 (16:44 +0000)]
Handle cases where a frame may have no data. (issue #9519 reported by dmb)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66437
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Olle Johansson [Tue, 29 May 2007 16:07:44 +0000 (16:07 +0000)]
Don't reset hangupcause if we already have one
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66414
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Olle Johansson [Tue, 29 May 2007 16:02:50 +0000 (16:02 +0000)]
Tracking down hanging channels, killing them one by one. Issue #9235 and related
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66404
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Joshua Colp [Tue, 29 May 2007 15:43:16 +0000 (15:43 +0000)]
Update datastores documentation. (issue #9801 reported by mnicholson)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66398
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Olle Johansson [Tue, 29 May 2007 09:41:40 +0000 (09:41 +0000)]
Merged revisions 66349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines
Issue #9802 - Change inuse counter on CANCEL
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Joshua Colp [Mon, 28 May 2007 23:16:56 +0000 (23:16 +0000)]
Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau)
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Jason Parker [Sun, 27 May 2007 04:12:37 +0000 (04:12 +0000)]
I don't know what this was trying to do, but it's clearly incorrect.
Issues 9808 and 9809.
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Kevin P. Fleming [Fri, 25 May 2007 14:43:54 +0000 (14:43 +0000)]
have to check for OSP toolkit _after_ checking for OpenSSL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66160
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Tilghman Lesher [Fri, 25 May 2007 14:41:27 +0000 (14:41 +0000)]
Merged revisions 66127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) | 2 lines
Issue 9791 - Fix pronunciation of seconds in Dutch
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Kevin P. Fleming [Fri, 25 May 2007 14:28:46 +0000 (14:28 +0000)]
handle the GNUTLS library properly in the configure script and build system
don't build in OSP support unless we have found and are allowed to use SSL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66157
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Russell Bryant [Thu, 24 May 2007 22:23:59 +0000 (22:23 +0000)]
if the string field init fails, clean up the stuff that was allocated already
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66076
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Joshua Colp [Thu, 24 May 2007 22:16:58 +0000 (22:16 +0000)]
Fix slinfactory logic when dealing with frames coming in that may already be in the signed linear format.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66074
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Russell Bryant [Thu, 24 May 2007 22:07:39 +0000 (22:07 +0000)]
Check the result of ast_string_field_init() in ast_channel_alloc()
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66070
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Russell Bryant [Thu, 24 May 2007 22:06:13 +0000 (22:06 +0000)]
Make 1.4 build on my machine, too..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66068
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Jason Parker [Thu, 24 May 2007 20:54:16 +0000 (20:54 +0000)]
Rebuild configure script for previous ar fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66030
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Jason Parker [Thu, 24 May 2007 20:53:18 +0000 (20:53 +0000)]
Following moving strip to AC_PATH_TOOL, we need to do something similar for ar.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66029
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Russell Bryant [Thu, 24 May 2007 20:42:53 +0000 (20:42 +0000)]
Checking for the strip application needs to be done with AC_PATH_TOOL
instead of AC_PATH_PROG to properly handle cross compilation environments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66026
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Russell Bryant [Thu, 24 May 2007 19:05:08 +0000 (19:05 +0000)]
Clear CFLAGS before running make for menuselect.
(issue #9784, reported by ovi, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65978
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Kevin P. Fleming [Thu, 24 May 2007 18:28:48 +0000 (18:28 +0000)]
oops, use #ifdef instead of #if
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65967
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Kevin P. Fleming [Thu, 24 May 2007 18:25:21 +0000 (18:25 +0000)]
don't reference GnuTLS headers and functions unless the configure script found it
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65966
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Kevin P. Fleming [Thu, 24 May 2007 18:24:55 +0000 (18:24 +0000)]
don't use uninitialized variables
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65965
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Joshua Colp [Thu, 24 May 2007 15:27:23 +0000 (15:27 +0000)]
Add the ability to blacklist certain commands from being executed using the Command AMI action. (issue #9240 reported by junky)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65902
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Olle Johansson [Thu, 24 May 2007 15:26:10 +0000 (15:26 +0000)]
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65901
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Olle Johansson [Thu, 24 May 2007 15:20:54 +0000 (15:20 +0000)]
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65892
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Jason Parker [Thu, 24 May 2007 15:16:52 +0000 (15:16 +0000)]
Update cleancount for that last commit - just for good measure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65883
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Jason Parker [Thu, 24 May 2007 15:14:02 +0000 (15:14 +0000)]
Fix handling of zero-length frames when a codec is capable of native PLC.
Issue 9183, patch by Mihai.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65877
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Dwayne M. Hubbard [Thu, 24 May 2007 15:08:56 +0000 (15:08 +0000)]
merged qwell's func_math patch for issue 9507
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65866
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Joshua Colp [Thu, 24 May 2007 15:08:17 +0000 (15:08 +0000)]
I like it when the RTP stack compiles myself...
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Olle Johansson [Thu, 24 May 2007 15:05:10 +0000 (15:05 +0000)]
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65857
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Russell Bryant [Thu, 24 May 2007 15:04:14 +0000 (15:04 +0000)]
Ensure that frames are fully initialized. This will probably fix getting
weird timestamp log messages in logs when using the Festival app.
(issue #9781, patch by me)
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Russell Bryant [Thu, 24 May 2007 14:49:05 +0000 (14:49 +0000)]
Fix the calculation of the RTT for RTCP. The previous code would result in
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65842
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Olle Johansson [Thu, 24 May 2007 14:48:55 +0000 (14:48 +0000)]
Issue #8536 - Caller ID not set in CDR for jingle
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Joshua Colp [Thu, 24 May 2007 14:42:12 +0000 (14:42 +0000)]
Merged revisions 65837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines
Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
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Olle Johansson [Thu, 24 May 2007 14:38:30 +0000 (14:38 +0000)]
Issue 8409 - phsultan - Fix "login" as component to jabber server.
...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
of BYE requests.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836
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Christian Richter [Thu, 24 May 2007 09:37:32 +0000 (09:37 +0000)]
Merged revisions 65767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line
we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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Kevin P. Fleming [Wed, 23 May 2007 20:59:19 +0000 (20:59 +0000)]
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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Kevin P. Fleming [Wed, 23 May 2007 20:51:56 +0000 (20:51 +0000)]
Merged revisions 65682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines
ensure that variables are set on a newly created channel before we start a PBX on it
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Kevin P. Fleming [Wed, 23 May 2007 20:35:50 +0000 (20:35 +0000)]
clear the 'delay PBX' flag when we are ready to start the PBX
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Kevin P. Fleming [Wed, 23 May 2007 20:30:24 +0000 (20:30 +0000)]
don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
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Kevin P. Fleming [Wed, 23 May 2007 20:07:59 +0000 (20:07 +0000)]
Merged revisions 65676 via svnmerge from
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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines
if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it
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Russell Bryant [Wed, 23 May 2007 13:07:13 +0000 (13:07 +0000)]
Merged revisions 65588 via svnmerge from
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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines
Revert revision 62417 as someone reported problems with it to Mark. This was
related to issue #9588.
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Kevin P. Fleming [Tue, 22 May 2007 20:25:41 +0000 (20:25 +0000)]
when building a version string for a developer branch, include the base branch in the version string
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Russell Bryant [Tue, 22 May 2007 18:40:38 +0000 (18:40 +0000)]
List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)
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Joshua Colp [Tue, 22 May 2007 15:04:46 +0000 (15:04 +0000)]
Remove a double const.
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BJ Weschke [Tue, 22 May 2007 14:02:56 +0000 (14:02 +0000)]
Fix a problem with flag recognition.
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Russell Bryant [Tue, 22 May 2007 13:09:34 +0000 (13:09 +0000)]
Merged revisions 65389 via svnmerge from
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r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | 4 lines
Fix a memory leak that I just noticed in the device state handling in app_queue.
On most device state changes, it would leak roughly 8 to 64 bytes (the length of
the name of the device).
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Christian Richter [Tue, 22 May 2007 08:12:20 +0000 (08:12 +0000)]
Merged revisions 65328 via svnmerge from
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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line
we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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Joshua Colp [Sun, 20 May 2007 17:59:58 +0000 (17:59 +0000)]
res_agi needs to export two symbols (ast_agi_register and ast_agi_unregister) for usage by others. (issue #9755 reported by mnicholson)
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Steve Murphy [Fri, 18 May 2007 22:26:51 +0000 (22:26 +0000)]
Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call
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Steve Murphy [Fri, 18 May 2007 22:06:27 +0000 (22:06 +0000)]
Merged revisions 65172 via svnmerge from
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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Olle Johansson [Fri, 18 May 2007 18:16:09 +0000 (18:16 +0000)]
Merged revisions 65122 via svnmerge from
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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
Not getting an ACK to a 200 OK in the initial invite is critical to the call.
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Olle Johansson [Fri, 18 May 2007 15:18:13 +0000 (15:18 +0000)]
Merged revisions 65075 via svnmerge from
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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
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Olle Johansson [Fri, 18 May 2007 12:58:39 +0000 (12:58 +0000)]
- Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp.
- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.
Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.
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Christian Richter [Fri, 18 May 2007 12:40:46 +0000 (12:40 +0000)]
Merged revisions 65007 via svnmerge from
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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line
fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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Olle Johansson [Fri, 18 May 2007 10:37:44 +0000 (10:37 +0000)]
Issue 9487 - stop media flows at hangup of call
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Christian Richter [Fri, 18 May 2007 08:58:51 +0000 (08:58 +0000)]
Merged revisions 64902 via svnmerge from
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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line
we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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Russell Bryant [Fri, 18 May 2007 02:48:51 +0000 (02:48 +0000)]
Fix a small bug I noticed while working on something else. app_queue did not
unregister its device state monitoring callback in unload_module(). So, this
would make Asterisk crash on the first device state change after you
unload the module.
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Tilghman Lesher [Thu, 17 May 2007 21:19:34 +0000 (21:19 +0000)]
Merged revisions 64819 via svnmerge from
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r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines
How is it that we never caught that this is returning the opposite of our documentation, until now?
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Jason Parker [Thu, 17 May 2007 16:53:27 +0000 (16:53 +0000)]
Merged revisions 64758 via svnmerge from
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r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 lines
If we have a negative current message, we shouldn't go back even further...
Issue 9727.
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Russell Bryant [Thu, 17 May 2007 16:52:53 +0000 (16:52 +0000)]
Remove script that is no longer functional since the build system was redone.
(issue #9340, reported by junky)
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Russell Bryant [Thu, 17 May 2007 16:47:29 +0000 (16:47 +0000)]
Increase the size of a buffer to support longer dial strings for channels.
(issue #9291, reported and fix suggested by meni)
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Joshua Colp [Thu, 17 May 2007 16:10:12 +0000 (16:10 +0000)]
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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Joshua Colp [Thu, 17 May 2007 13:48:44 +0000 (13:48 +0000)]
Fix authuser support. (issue #9740 reported by xmarksthespot)
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Russell Bryant [Thu, 17 May 2007 06:13:53 +0000 (06:13 +0000)]
Update the main README to reflect the new build process for 1.4 and above.
(issue #9725, patch by eliel)
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Olle Johansson [Wed, 16 May 2007 11:01:11 +0000 (11:01 +0000)]
Blocking patch already in this code
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Olle Johansson [Wed, 16 May 2007 10:59:28 +0000 (10:59 +0000)]
Fix auth on BYE. (Different patch than for 1.2)
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Olle Johansson [Wed, 16 May 2007 10:38:18 +0000 (10:38 +0000)]
Issue #9681 - Handle www-auth on BYE
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Olle Johansson [Wed, 16 May 2007 10:05:47 +0000 (10:05 +0000)]
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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Olle Johansson [Wed, 16 May 2007 09:57:22 +0000 (09:57 +0000)]
Issue #9439 - properly handle username parameters in SIP uri.
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Olle Johansson [Wed, 16 May 2007 09:12:34 +0000 (09:12 +0000)]
Merged revisions 64535 via svnmerge from
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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines
Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)
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Olle Johansson [Wed, 16 May 2007 08:46:18 +0000 (08:46 +0000)]
Merged following patch with a lot of changes for 1.4
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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines
Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that are not supposed to register.
My patch, stole the issue report from Russell. My apologies, Russell :-)
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Christian Richter [Wed, 16 May 2007 08:44:51 +0000 (08:44 +0000)]
Merged revisions 64513 via svnmerge from
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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Russell Bryant [Tue, 15 May 2007 19:52:18 +0000 (19:52 +0000)]
Properly fix a problem that occurs when you set PARKINGEXTEN to an exten where
a call is already parked. (issue #9723, patch by me)
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Russell Bryant [Mon, 14 May 2007 21:16:39 +0000 (21:16 +0000)]
When someone requests a specific parking space using the PARKINGEXTEN variable,
ensure that no other caller is already there.
(issue #9723, reported by mdu113, patch by me)
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