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git.ipfire.org Git - thirdparty/asterisk.git/log
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
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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...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65863
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65853
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679
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Kevin P. Fleming [Wed, 23 May 2007 20:07:59 +0000 (20:07 +0000)]
Merged revisions 65676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65541
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65408
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Russell Bryant [Tue, 22 May 2007 13:09:34 +0000 (13:09 +0000)]
Merged revisions 65389 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65250
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041
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Christian Richter [Fri, 18 May 2007 12:40:46 +0000 (12:40 +0000)]
Merged revisions 65007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64868
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Tilghman Lesher [Thu, 17 May 2007 21:19:34 +0000 (21:19 +0000)]
Merged revisions 64819 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64756
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754
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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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Olle Johansson [Mon, 14 May 2007 19:26:50 +0000 (19:26 +0000)]
Change -2 to XMIT_ERROR to clarify a bit more
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Russell Bryant [Mon, 14 May 2007 19:13:00 +0000 (19:13 +0000)]
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.
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Olle Johansson [Mon, 14 May 2007 18:52:09 +0000 (18:52 +0000)]
Handle network errors, like host or network unreachable, in a better way. This means that
calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.
Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.
This is already implemented in trunk.
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Joshua Colp [Mon, 14 May 2007 18:48:33 +0000 (18:48 +0000)]
Properly set datalen field when doing PLC in codec_speex. (issue #9722 reported by mihai)
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Joshua Colp [Mon, 14 May 2007 18:36:34 +0000 (18:36 +0000)]
Merged revisions 64275 via svnmerge from
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r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 lines
Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies)
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Joshua Colp [Mon, 14 May 2007 17:23:51 +0000 (17:23 +0000)]
Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold.
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Steve Murphy [Mon, 14 May 2007 13:58:42 +0000 (13:58 +0000)]
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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Olle Johansson [Mon, 14 May 2007 10:39:12 +0000 (10:39 +0000)]
Add hangupcause when we lack codecs for transcoding
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Joshua Colp [Sat, 12 May 2007 22:27:04 +0000 (22:27 +0000)]
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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Joshua Colp [Sat, 12 May 2007 21:10:45 +0000 (21:10 +0000)]
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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Joshua Colp [Sat, 12 May 2007 16:32:15 +0000 (16:32 +0000)]
Ensure the onhold flag is set no matter what when being put on hold.
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Jason Parker [Fri, 11 May 2007 20:16:17 +0000 (20:16 +0000)]
Hide manager password from "manager show user foo".
I realize that there are other ways to get this,
but we really don't need to just show it in plain text so easily.
Issue 9273, patch by junky
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Tilghman Lesher [Fri, 11 May 2007 16:35:51 +0000 (16:35 +0000)]
Merged revisions 63903 via svnmerge from
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r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) | 2 lines
Issue 9121 - fixups for safe_asterisk script
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Russell Bryant [Fri, 11 May 2007 16:05:43 +0000 (16:05 +0000)]
When MD5 authentication is not possible because there is no challenge present,
either because the Challenge action was never issued, or some other reason,
give a proper error message and return an error instead of claiming that the
user wasn't found.
(reported by jsmith on IRC)
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Joshua Colp [Fri, 11 May 2007 15:43:14 +0000 (15:43 +0000)]
Make the PARKINGEXTEN feature of parking actually work. (issue #9708 reported by mdu113)
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Jason Parker [Thu, 10 May 2007 23:15:37 +0000 (23:15 +0000)]
Merged revisions 63828 via svnmerge from
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r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines
Fix an issue with trying to kill a thread before it gets created.
Issue 9709, patch by nic_bellamy.
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Russell Bryant [Thu, 10 May 2007 22:23:42 +0000 (22:23 +0000)]
Strip terminal escape sequences from CLI command output that is going to be
sent out over the manager interface.
(issue #9659, reported by pari, fixed by me)
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Doug Bailey [Thu, 10 May 2007 20:48:54 +0000 (20:48 +0000)]
Add test for negative offsets in cid data to prevent infinite loops.
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Olle Johansson [Thu, 10 May 2007 20:46:41 +0000 (20:46 +0000)]
Merged revisions 63748 via svnmerge from
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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines
Do not allocate SIP pvt's for PEERs we can not reach.
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
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Joshua Colp [Wed, 9 May 2007 19:22:39 +0000 (19:22 +0000)]
Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.
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Joshua Colp [Wed, 9 May 2007 17:43:30 +0000 (17:43 +0000)]
Do not prematurely go on hold if sendonly was not actually set.
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Matthew Fredrickson [Wed, 9 May 2007 17:25:21 +0000 (17:25 +0000)]
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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Russell Bryant [Wed, 9 May 2007 16:55:27 +0000 (16:55 +0000)]
Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the
code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events. (pointed out by Michael Neuhauser on the
asterisk-dev list)
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Joshua Colp [Wed, 9 May 2007 16:54:56 +0000 (16:54 +0000)]
Merged revisions 63610 via svnmerge from
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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Russell Bryant [Wed, 9 May 2007 16:43:50 +0000 (16:43 +0000)]
Only call ast_senddigit_begin() in ast_senddigit() if the channel has a
send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.
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Tilghman Lesher [Wed, 9 May 2007 14:50:33 +0000 (14:50 +0000)]
Merged revisions 63565 via svnmerge from
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r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) | 2 lines
Replicate fix from 51158 (app_voicemail) to app_directory (Issue 9224)
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Russell Bryant [Wed, 9 May 2007 13:24:03 +0000 (13:24 +0000)]
I have seen multiple people post questions trying to figure out what the
message "The configure script must be executed before running 'make'" means.
So, add another like that says to specifically run ./configure. If this isn't
obvious enough, then they should be using something like AsteriskNOW and not
installing from source.
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Christian Richter [Wed, 9 May 2007 13:17:10 +0000 (13:17 +0000)]
Merged revisions 62945,63402,63519 via svnmerge from
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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Olle Johansson [Wed, 9 May 2007 13:04:14 +0000 (13:04 +0000)]
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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Tilghman Lesher [Tue, 8 May 2007 22:38:02 +0000 (22:38 +0000)]
Merged revisions 63477 via svnmerge from
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r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) | 2 lines
Issue 9602 - segfault in app_macro
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Russell Bryant [Tue, 8 May 2007 16:53:09 +0000 (16:53 +0000)]
I mixed up the use of the find_feature() function, so I renamed it
find_dynamic_feature, and changed the code to use the correct lock when
using it.
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Russell Bryant [Tue, 8 May 2007 16:30:43 +0000 (16:30 +0000)]
Use a read/write lock when accessing the built-in features.
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Russell Bryant [Tue, 8 May 2007 15:10:37 +0000 (15:10 +0000)]
Move realtime_pgsql.sql to contrib/scripts to be with the rest of the sql examples.
(issue #9676, suretec)
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Tilghman Lesher [Tue, 8 May 2007 06:22:37 +0000 (06:22 +0000)]
Merged revisions 63359 via svnmerge from
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r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) | 2 lines
Issue 9527 - upon entering a folder, no message is selected (curmsg == -1), so deleting causes memory corruption (beyond bounds)
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