Russell Bryant
5272289355
I thought I noticed a memory leak earlier when I saw that the contents of this
...
list were not destroyed when the module is unloaded. However, after reading
the code related to the use of this list a lot today, I realized that it isn't
necessary. So, I have added a comment to explain why it isn't necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-18 22:52:54 +00:00
Tilghman Lesher
9c99cba373
Change IAX variables to use datastores (closes issue #9315 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-18 22:40:03 +00:00
Russell Bryant
4681316959
Merged revisions 75759 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75759 | russell | 2007-07-18 16:09:46 -0500 (Wed, 18 Jul 2007) | 13 lines
Merged revisions 75757 via svnmerge from
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r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | 5 lines
When traversing the queue of frames for possible retransmission after
receiving a VNAK, handle sequence number wraparound so that all frames that
should be retransmitted actually do get retransmitted.
(issue #10227 , reported and patched by mihai)
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2007-07-18 21:10:27 +00:00
Jason Parker
89acafbf37
Merged revisions 75732 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul 2007) | 1 line
Umm, why are we transmitting dialtone on cfwdall?
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2007-07-18 20:18:00 +00:00
Jason Parker
4f22126365
Merged revisions 75711 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #9245 )
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r75711 | qwell | 2007-07-18 14:54:32 -0500 (Wed, 18 Jul 2007) | 4 lines
Fixes for 7935/7936 conference phones.
Issue 9245, patch by slimey.
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2007-07-18 20:05:45 +00:00
Jason Parker
a1b2a23b4d
Merged revisions 75707 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #9887 )
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r75707 | qwell | 2007-07-18 14:48:12 -0500 (Wed, 18 Jul 2007) | 4 lines
Fix issues with new 79x1 phones.
Issue 9887, patches by DEA
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2007-07-18 19:51:18 +00:00
Russell Bryant
707ca59923
convert some lines indented with spaces to tabs
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2007-07-18 19:50:21 +00:00
Tilghman Lesher
81bc1d7af5
Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second
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2007-07-18 19:47:20 +00:00
Joshua Colp
a23feea9d2
Merged revisions 75623 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines
Few more places that needs to check for onhold state.
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2007-07-18 15:45:18 +00:00
Joshua Colp
d90bddfa6c
Merged revisions 75621 via svnmerge from
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r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 lines
(closes issue #10165 )
Reported by: elandivar
It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.
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2007-07-18 15:42:11 +00:00
Joshua Colp
e2b0f321a3
Merged revisions 75619 via svnmerge from
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r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2 lines
Don't bother reloading chan_h323 if it did not load successfully in the first place. This would otherwise cause a crash.
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2007-07-18 15:26:52 +00:00
Steve Murphy
5ac24b25d3
This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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2007-07-18 14:35:07 +00:00
Joshua Colp
4003b31fc5
Minor code tweaks. Variables were being checked wrong in some situations and didn't need to be checked in others.
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2007-07-18 13:59:01 +00:00
Russell Bryant
91e2afe4e7
Merged revisions 75450 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75450 | russell | 2007-07-17 15:57:56 -0500 (Tue, 17 Jul 2007) | 11 lines
Merged revisions 75449 via svnmerge from
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r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 Jul 2007) | 3 lines
Properly check for the length in the skinny packet to prevent an invalid memcpy.
(ASA-2007-016)
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2007-07-17 20:58:40 +00:00
Russell Bryant
b5ff588044
Merged revisions 75445 via svnmerge from
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r75445 | russell | 2007-07-17 15:48:21 -0500 (Tue, 17 Jul 2007) | 13 lines
Merged revisions 75444 via svnmerge from
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r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | 5 lines
Ensure that when encoding the contents of an ast_frame into an iax_frame, that
the size of the destination buffer is known in the iax_frame so that code
won't write past the end of the allocated buffer when sending outgoing frames.
(ASA-2007-014)
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2007-07-17 20:49:09 +00:00
Russell Bryant
681e45f0b2
Merged revisions 75441 via svnmerge from
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r75441 | russell | 2007-07-17 15:42:12 -0500 (Tue, 17 Jul 2007) | 12 lines
Merged revisions 75440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | 4 lines
After parsing information elements in IAX frames, set the data length to zero,
so that code later on does not think it has data to copy.
(ASA-2007-015)
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2007-07-17 20:42:50 +00:00
Steve Murphy
8a7732f067
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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2007-07-17 19:40:29 +00:00
Russell Bryant
432a087ed9
Merged revisions 75053 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75053 | russell | 2007-07-13 14:11:26 -0500 (Fri, 13 Jul 2007) | 20 lines
Merged revisions 75052 via svnmerge from
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r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines
(closes issue #9660 )
Reported by: mmacvicar
Patches submitted by: bbryant, russell
Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous
When using a TDM400P (and probably other analog cards) there was a chance that
you could hang up and pick the phone back up where it has been long enough to
be not considered a flash hook, but too soon such that the device reports that
it is busy and the person on the phone will only hear silence. This patch
makes chan_zap more tolerant of this and gives the device a couple of seconds
to succeed so the person on the phone happily gets their dialtone.
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2007-07-13 19:19:51 +00:00
Mark Michelson
4d915d13c7
Merged revisions 74997 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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2007-07-12 23:02:33 +00:00
Steve Murphy
6bc0a4929c
Merged revisions 74955 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 line
This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up
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2007-07-12 20:46:32 +00:00
Jason Parker
4cc5a034b7
Merged revisions 74866 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul 2007) | 1 line
It helps if I actually add this stuff for the 7921 too - otherwise it won't actually do much of anything.
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2007-07-12 16:54:02 +00:00
Jason Parker
588c3f85fe
Merged revisions 74864 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul 2007) | 1 line
Add device ID for 7921 wireless skinny phone
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2007-07-12 16:49:17 +00:00
Russell Bryant
08b1a342e5
Merged revisions 74767 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74767 | russell | 2007-07-11 17:57:07 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines
The function make_trunk() can fail and return -1 instead of a valid new call
number. Fix the uses of this function to handle this instead of treating it
as the new call number. This would cause a deadlock and memory corruption.
(possible cause of issue #9614 and others, patch by me)
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2007-07-11 23:05:34 +00:00
Mark Michelson
66449656d3
Merged revisions 74722 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74722 | mmichelson | 2007-07-11 16:14:09 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 Jul 2007) | 5 lines
The cli command "agent logoff Agent/x soft" did not work...at all. Now it does.
(closes issue #10178 , reported and patched by makoto, with slight modification for 1.4 and trunk by me)
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2007-07-11 21:15:47 +00:00
Joshua Colp
2d3f9ce25b
Merged revisions 74572 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2 lines
Instead of figuring out kernel versions that have compiler.h and not... let's just use autoconf to check for it's presence. (issue #10174 reported by francesco_r)
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2007-07-11 17:06:54 +00:00
Joshua Colp
2162433060
Change header a bit to get rid of a doxygen parse error. (issue #10177 reported by snuffy)
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2007-07-11 14:38:01 +00:00
Joshua Colp
2df8b91be1
Merged revisions 74515 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul 2007) | 2 lines
Only check if we need to do a SIGMA based tone generation if we have a card. (issue #10179 reported by mikowhy)
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2007-07-11 14:10:38 +00:00
Jason Parker
d459994f7c
Merged revisions 74379 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10169 )
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r74379 | qwell | 2007-07-10 14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines
Merged revisions 74376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 lines
Fix an issue with wrapuptime not working when using AgentLogin.
Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618
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2007-07-10 19:07:25 +00:00
Jason Parker
76e2e39f77
Fix building that was broken by recent monitor.h changes. Thanks Russell for pointing this out (and pointing out what I probably did to prevent gcc from fixing it - don't ctrl-C builds)
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2007-07-10 15:07:25 +00:00
Jason Parker
d61b0cac84
Merged revisions 74159 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
Closes issue #9186
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r74159 | qwell | 2007-07-09 15:19:28 -0500 (Mon, 09 Jul 2007) | 16 lines
Merged revisions 74158 via svnmerge from
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r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines
Several chan_zap options were not working on reload because they were arbitrarily
disallowed when reloading some/most PRI options (such as signalling) was disallowed.
Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload.
This corrects that behavior.
Issue 9186, patch by tzafrir.
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2007-07-09 20:21:13 +00:00
Russell Bryant
e4f28bc1a5
remove an unused variable
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2007-07-09 18:58:09 +00:00
Joshua Colp
ec8ef98a03
Merged revisions 74082 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul 2007) | 2 lines
Only destroy the scheduler context if it was allocated. (issue #10124 reported by gzero)
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2007-07-09 15:34:54 +00:00
Joshua Colp
836ea36d3c
Merged revisions 74045 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2 lines
Few minor thread synchronization tweaks. (issue #10124 reported by gzero)
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2007-07-09 14:50:04 +00:00
Olle Johansson
a1b9cbcd31
Implementation of a feature that will disable "missed calls" counters on SIP phones.
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If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-09 08:27:37 +00:00
Tilghman Lesher
ba857cc8a9
Merged revisions 73985 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines
Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104 )
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2007-07-09 04:09:16 +00:00
Olle Johansson
74e8ab14fc
Merged revisions 73849 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 lines
While tracking down a bug, I need some more history. Dumphistory is very useful, indeed.
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2007-07-08 09:49:21 +00:00
Russell Bryant
1da115c8d9
Merged revisions 73769 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r73769 | russell | 2007-07-06 18:02:58 -0500 (Fri, 06 Jul 2007) | 12 lines
Merged revisions 73768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines
If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one. If this isn't done, Asterisk
will crash. (issue #10120 )
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2007-07-06 23:05:24 +00:00
Russell Bryant
a0c37d2548
Merged revisions 73679 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r73679 | russell | 2007-07-06 10:57:25 -0500 (Fri, 06 Jul 2007) | 15 lines
Merged revisions 73678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines
(closes issue #10125 )
Reported by: makoto
Patches submitted by: makoto
This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.
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2007-07-06 16:00:03 +00:00
Kevin P. Fleming
8b65011334
it really seems pointless to run gentone to create these header files every time we build Asterisk...
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2007-07-06 15:47:16 +00:00
Mark Michelson
1745732a37
Merged revisions 73675 via svnmerge from
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r73675 | mmichelson | 2007-07-06 10:27:28 -0500 (Fri, 06 Jul 2007) | 13 lines
Merged revisions 73674 via svnmerge from
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r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 Jul 2007) | 5 lines
Fixed a bug wherein agents get stuck busy. (issue 9618, reported by jiddings, patched by moi)
closes issue #9618
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2007-07-06 15:28:28 +00:00
Russell Bryant
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r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) | 3 lines
Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never
started. (closes issue #10124 , reported by gzero, fixed by me)
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2007-07-05 23:59:50 +00:00
Russell Bryant
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r73555 | russell | 2007-07-05 18:05:33 -0500 (Thu, 05 Jul 2007) | 3 lines
copy from the correct buffer when deferring a full frame
(related to issue #9937 )
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2007-07-05 23:06:01 +00:00
Russell Bryant
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r73551 | russell | 2007-07-05 17:31:31 -0500 (Thu, 05 Jul 2007) | 6 lines
* Store the call number that a thread is processing without the full frame bit
set to ease debugging
* When deferring a full frame for processing, stick it into the queue for the
thread that is processing frames for that call, not the one that read the
current frame and is about to go back into the idle list
(related to issue #9937 )
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2007-07-05 22:32:30 +00:00
Kevin P. Fleming
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r73548 | kpfleming | 2007-07-05 17:20:44 -0500 (Thu, 05 Jul 2007) | 10 lines
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r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines
we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729
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2007-07-05 22:29:37 +00:00
Joshua Colp
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r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, 05 Jul 2007) | 10 lines
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r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines
Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)
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2007-07-05 19:20:12 +00:00
Russell Bryant
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r73398 | russell | 2007-07-05 10:28:27 -0500 (Thu, 05 Jul 2007) | 2 lines
Make this module build for me in dev-mode
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2007-07-05 15:29:24 +00:00
Joshua Colp
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r73319 | file | 2007-07-05 10:27:40 -0300 (Thu, 05 Jul 2007) | 10 lines
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r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines
Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto)
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2007-07-05 13:28:41 +00:00
Christian Richter
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added general Jitterbuffer Implementation. #9960
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2007-07-05 07:45:21 +00:00
Christian Richter
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r73253 | crichter | 2007-07-04 16:53:48 +0200 (Mi, 04 Jul 2007) | 9 lines
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r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 Jul 2007) | 1 line
bchannel configurations like echocancel and volume control, need to be setuped on inbound calls too.
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2007-07-04 14:58:58 +00:00
Christian Richter
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r73208 | crichter | 2007-07-04 10:27:44 +0200 (Mi, 04 Jul 2007) | 9 lines
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r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 Jul 2007) | 1 line
bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed..
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2007-07-04 08:43:27 +00:00