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git.ipfire.org Git - thirdparty/asterisk.git/log
Joshua Colp [Tue, 20 Feb 2007 23:57:03 +0000 (23:57 +0000)]
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55717
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Joshua Colp [Tue, 20 Feb 2007 23:08:45 +0000 (23:08 +0000)]
Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska)
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Joshua Colp [Tue, 20 Feb 2007 22:47:00 +0000 (22:47 +0000)]
Merged revisions 55669 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
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Russell Bryant [Tue, 20 Feb 2007 20:26:06 +0000 (20:26 +0000)]
Add the Asterisk version information to the Server header in HTTP
responses. (requested by Pari)
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Russell Bryant [Tue, 20 Feb 2007 19:57:07 +0000 (19:57 +0000)]
Increase the maximum number of manager headers to 128, at the request of Pari.
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Russell Bryant [Tue, 20 Feb 2007 19:53:15 +0000 (19:53 +0000)]
Blocked revisions 55588 via svnmerge
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r55588 | russell | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines
Convert a tab to spaces so that the documentation is printed out properly
aligned.
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Jason Parker [Tue, 20 Feb 2007 16:53:45 +0000 (16:53 +0000)]
No need to cast nor free with strdupa (thanks file)
55555!
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Russell Bryant [Tue, 20 Feb 2007 16:41:57 +0000 (16:41 +0000)]
Change the formatting of sla.conf.sample to make it more readable.
(issue #9112, blitzrage)
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Olle Johansson [Mon, 19 Feb 2007 21:12:55 +0000 (21:12 +0000)]
- Not sending arguments to an application is not "out of memory"
- Making error messages a bit more clear
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55483
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Tilghman Lesher [Mon, 19 Feb 2007 18:11:48 +0000 (18:11 +0000)]
Merged revisions 55434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) | 2 lines
forcename and forcegreetings options should check to see if the recording already exists
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Doug Bailey [Mon, 19 Feb 2007 14:52:59 +0000 (14:52 +0000)]
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55397
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Olle Johansson [Sun, 18 Feb 2007 12:35:54 +0000 (12:35 +0000)]
Merged revisions 55277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines
Documentation update (#9053, jsmith)
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Olle Johansson [Sun, 18 Feb 2007 10:51:22 +0000 (10:51 +0000)]
Block patch that was made only for 1.2 (already implemented in 1.4 and trunk)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55250
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Joshua Colp [Sat, 17 Feb 2007 17:39:32 +0000 (17:39 +0000)]
Add missing membername option to AddQueueMember documentation. (issue #9088 reported by seanbright)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55219
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Jason Parker [Sat, 17 Feb 2007 17:10:09 +0000 (17:10 +0000)]
Fix an issue where callerid would not be displayed on some phones.
Issue 8995, initial patch and research done by wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55217
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Joshua Colp [Sat, 17 Feb 2007 03:55:30 +0000 (03:55 +0000)]
Merged revisions 55153 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Answer the channel before recording privacy information. (issue #8926 reported by lmamane)
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Joshua Colp [Sat, 17 Feb 2007 02:59:50 +0000 (02:59 +0000)]
Make the 'i' option of Queue actually work. (issue #8986 reported by utis)
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Joshua Colp [Sat, 17 Feb 2007 01:16:59 +0000 (01:16 +0000)]
Merged revisions 55073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)
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Russell Bryant [Sat, 17 Feb 2007 00:40:34 +0000 (00:40 +0000)]
If the pg_config application is found, but there is probably executing it,
then consider postgres unavailable. (issue #8637)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55052
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Russell Bryant [Sat, 17 Feb 2007 00:31:42 +0000 (00:31 +0000)]
Filter out yet another architecture that does not work with the optimizations
in the built-in libgsm. (issue 8637, ovi)
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Russell Bryant [Fri, 16 Feb 2007 22:49:42 +0000 (22:49 +0000)]
Merged revisions 55005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
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Joshua Colp [Fri, 16 Feb 2007 22:18:46 +0000 (22:18 +0000)]
Merged revisions 54999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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Russell Bryant [Fri, 16 Feb 2007 21:12:18 +0000 (21:12 +0000)]
Merged revisions 54955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines
For conferences that are configured in meetme.conf, check the configuration
file every time someone joins the conference instead of only when the
conference is first created. This is to ensure that changes to the pin
numbers in the config file are always honored. (issue #9073)
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Joshua Colp [Fri, 16 Feb 2007 18:51:15 +0000 (18:51 +0000)]
Need to check macro extension as well as macro context for directed pickup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54924
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Russell Bryant [Fri, 16 Feb 2007 18:03:41 +0000 (18:03 +0000)]
Fix setting "autofallthrough" to yes by default. It was set to enabled in
pbx.c. However, if the option was not present in extensions.conf, then
pbx_config.c would set it back to disabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54898
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Russell Bryant [Fri, 16 Feb 2007 17:40:38 +0000 (17:40 +0000)]
Clean up a few coding guidelines issues - spaces to tabs, use sizeof() to pass
the size of a static buffer, add spaces ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54888
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Jason Parker [Fri, 16 Feb 2007 17:25:21 +0000 (17:25 +0000)]
Clarify a restart message. It's silly, but the reporter had a very valid point.
Issue 9079
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54886
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Joshua Colp [Fri, 16 Feb 2007 17:02:35 +0000 (17:02 +0000)]
Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984 reported by jamesb63)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54884
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Olle Johansson [Fri, 16 Feb 2007 12:06:23 +0000 (12:06 +0000)]
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54787
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Olle Johansson [Fri, 16 Feb 2007 11:39:55 +0000 (11:39 +0000)]
Merged revisions 54771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 lines
Issue #9069 - If we open with TH we should not close with /TD. (seanbright)
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Joshua Colp [Fri, 16 Feb 2007 00:48:48 +0000 (00:48 +0000)]
Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54714
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Joshua Colp [Thu, 15 Feb 2007 16:19:39 +0000 (16:19 +0000)]
Merged revisions 54622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines
Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70)
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Joshua Colp [Wed, 14 Feb 2007 21:07:23 +0000 (21:07 +0000)]
Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54481
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Olle Johansson [Wed, 14 Feb 2007 18:44:58 +0000 (18:44 +0000)]
Block patch only needed in 1.2
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Matthew Fredrickson [Wed, 14 Feb 2007 16:56:40 +0000 (16:56 +0000)]
Merged revisions 54373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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Joshua Colp [Wed, 14 Feb 2007 01:09:40 +0000 (01:09 +0000)]
Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54290
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Russell Bryant [Tue, 13 Feb 2007 21:31:22 +0000 (21:31 +0000)]
Remove a couple of leftover debug messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54235
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Russell Bryant [Tue, 13 Feb 2007 20:56:50 +0000 (20:56 +0000)]
Fix the documentation on the return values from device state provider
registration and deletion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54218
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Russell Bryant [Tue, 13 Feb 2007 19:42:00 +0000 (19:42 +0000)]
If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54204
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Olle Johansson [Tue, 13 Feb 2007 18:41:15 +0000 (18:41 +0000)]
Blocking patch for 1.2 only
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Russell Bryant [Mon, 12 Feb 2007 19:17:08 +0000 (19:17 +0000)]
Change ast_set_state_callback() to ast_dial_set_state_callback()
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54103
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Russell Bryant [Mon, 12 Feb 2007 17:58:43 +0000 (17:58 +0000)]
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54066
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Joshua Colp [Mon, 12 Feb 2007 16:34:45 +0000 (16:34 +0000)]
Make the --without-oss argument work. (issue #9026 reported by puzzled)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54026
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Russell Bryant [Mon, 12 Feb 2007 15:38:39 +0000 (15:38 +0000)]
Fix a typo where "vmpassword" should be "vmsecret"
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Paul Cadach [Sat, 10 Feb 2007 09:09:49 +0000 (09:09 +0000)]
Fix VLDTMF reception
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53881
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Paul Cadach [Sat, 10 Feb 2007 09:08:55 +0000 (09:08 +0000)]
Much simpler than previous one ;-)
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Paul Cadach [Sat, 10 Feb 2007 09:07:11 +0000 (09:07 +0000)]
Provide correct DTMF duration
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Paul Cadach [Sat, 10 Feb 2007 09:04:47 +0000 (09:04 +0000)]
Bring deprecated 'debug channel <x|all>' command back
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Kevin P. Fleming [Sat, 10 Feb 2007 06:06:08 +0000 (06:06 +0000)]
don't display the --with-imap message unless --with-imap was specified without a path
use '-n' instead of '! -z' for tests
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Russell Bryant [Sat, 10 Feb 2007 01:02:00 +0000 (01:02 +0000)]
Add some output for "show application SLAStation/SLATrunk"
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Russell Bryant [Sat, 10 Feb 2007 00:41:57 +0000 (00:41 +0000)]
Change some text to properly state "On Hold", which was already done in trunk.
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Russell Bryant [Sat, 10 Feb 2007 00:35:09 +0000 (00:35 +0000)]
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810
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Russell Bryant [Sat, 10 Feb 2007 00:15:50 +0000 (00:15 +0000)]
When the Echo() application receives the digit '#', echo that back as well.
Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
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Kevin P. Fleming [Fri, 9 Feb 2007 23:52:44 +0000 (23:52 +0000)]
another dependency
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Kevin P. Fleming [Fri, 9 Feb 2007 23:51:41 +0000 (23:51 +0000)]
add some inter-module dependencies
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Kevin P. Fleming [Fri, 9 Feb 2007 23:51:29 +0000 (23:51 +0000)]
fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file
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Joshua Colp [Fri, 9 Feb 2007 19:33:31 +0000 (19:33 +0000)]
Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee)
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Kevin P. Fleming [Fri, 9 Feb 2007 16:42:22 +0000 (16:42 +0000)]
clarify the fact that voicemail IMAP storage cannot be built against a distro's binary c-client library package (at least not at this time)
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Olle Johansson [Thu, 8 Feb 2007 23:18:59 +0000 (23:18 +0000)]
Don't output debug unless we asked for it
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Joshua Colp [Thu, 8 Feb 2007 17:54:32 +0000 (17:54 +0000)]
Fix timeout issue when utterance is longer then timeout itself.
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Tilghman Lesher [Thu, 8 Feb 2007 13:47:54 +0000 (13:47 +0000)]
Issue 9007 - Mutex not released on early return
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Tilghman Lesher [Thu, 8 Feb 2007 13:40:02 +0000 (13:40 +0000)]
Merged revisions 53529 via svnmerge from
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r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) | 2 lines
Issue 9003 - If fullname is empty, quote() passes back "\""
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Russell Bryant [Wed, 7 Feb 2007 23:52:45 +0000 (23:52 +0000)]
When building libdb1.a, put the additional flags needed at the beginning of
ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
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Russell Bryant [Wed, 7 Feb 2007 20:07:39 +0000 (20:07 +0000)]
The clean target actually needs to run "distclean" on editline. This is
because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
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Joshua Colp [Wed, 7 Feb 2007 17:53:03 +0000 (17:53 +0000)]
We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982)
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Russell Bryant [Wed, 7 Feb 2007 17:39:31 +0000 (17:39 +0000)]
When parsing the NTP timestamp in a sender report message, you are supposed to
take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
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Joshua Colp [Wed, 7 Feb 2007 17:04:44 +0000 (17:04 +0000)]
Directly load say.conf in load_module instead of calling the reload function. (issue #8946 reported by junky)
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Joshua Colp [Wed, 7 Feb 2007 15:43:39 +0000 (15:43 +0000)]
Merged revisions 53357 via svnmerge from
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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines
Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst)
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Tilghman Lesher [Wed, 7 Feb 2007 15:33:51 +0000 (15:33 +0000)]
Merged revisions 53354 via svnmerge from
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r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) | 2 lines
Issue 7440 - Macro called from Macro from the h extension exits prematurely
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Christian Richter [Wed, 7 Feb 2007 09:22:44 +0000 (09:22 +0000)]
Merged revisions 52843 via svnmerge from
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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line
fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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Tilghman Lesher [Wed, 7 Feb 2007 05:24:31 +0000 (05:24 +0000)]
Text fix for jabber reload command (reported by bkruse via IRC)
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Tilghman Lesher [Tue, 6 Feb 2007 07:00:52 +0000 (07:00 +0000)]
Merged revisions 53245 via svnmerge from
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r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) | 2 lines
Issue 8987 - Status could return two responses (mnicholson)
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Olle Johansson [Mon, 5 Feb 2007 23:43:59 +0000 (23:43 +0000)]
Formatting
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Joshua Colp [Mon, 5 Feb 2007 17:06:18 +0000 (17:06 +0000)]
Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky)
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Joshua Colp [Mon, 5 Feb 2007 16:02:00 +0000 (16:02 +0000)]
Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky)
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Olle Johansson [Mon, 5 Feb 2007 00:18:34 +0000 (00:18 +0000)]
Add some comments on queue system behaviour and how it affects the
SIP channel
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Joshua Colp [Sat, 3 Feb 2007 21:05:02 +0000 (21:05 +0000)]
Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
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Russell Bryant [Sat, 3 Feb 2007 20:44:20 +0000 (20:44 +0000)]
Merged revisions 53133 via svnmerge from
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r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines
set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application
exits early because of invalid arguments instead of just leaving it empty.
(issue #8975)
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Russell Bryant [Sat, 3 Feb 2007 20:40:26 +0000 (20:40 +0000)]
Blocked revisions 53134 via svnmerge
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r53134 | russell | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines
Revert some changes that accidentally got committed as a part of another fix.
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Paul Cadach [Sat, 3 Feb 2007 10:02:55 +0000 (10:02 +0000)]
Remove quote from H.323 vendor string because due to compatibilities with CS1000 reported at www.voip-info.org
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BJ Weschke [Fri, 2 Feb 2007 21:26:45 +0000 (21:26 +0000)]
I'm baaaaaaaaaack. :)
Post a warning to the console that things might possibly be misconfigured when queue member's states are still 'Not in Use' when we're about to bridge them with a caller from queue. Also, put some documentation quoted from oej's queues.txt efforts started in /trunk today.
This commit puts #7433 into feedback state for 1.4, and pending no further negative feedback, it will finally be closed.
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Joshua Colp [Fri, 2 Feb 2007 17:15:22 +0000 (17:15 +0000)]
Correct a copy/pasted error message line for RTCP.
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Joshua Colp [Fri, 2 Feb 2007 16:59:53 +0000 (16:59 +0000)]
Merged revisions 53117 via svnmerge from
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r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 lines
Pass the glob expanded filename to process_text_line so that error messages contain the actual filename, not the original include one. (issue #8959 reported by tzafrir)
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Joshua Colp [Fri, 2 Feb 2007 15:29:35 +0000 (15:29 +0000)]
Add systemname to asterisk.conf generation per recent discussions about it. (issue #8968 reported by blitzrage)
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Olle Johansson [Fri, 2 Feb 2007 00:24:03 +0000 (00:24 +0000)]
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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Jason Parker [Thu, 1 Feb 2007 23:16:08 +0000 (23:16 +0000)]
Blocked revisions 53107 via svnmerge
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r53107 | qwell | 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix a small typo. Synopsis lines shouldn't have a newline
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Joshua Colp [Thu, 1 Feb 2007 22:24:32 +0000 (22:24 +0000)]
Merged revisions 53103 via svnmerge from
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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Joshua Colp [Thu, 1 Feb 2007 22:04:58 +0000 (22:04 +0000)]
Huh... fix the berkeley DB to compile here as well, but it apparently required both dev mode and no optimizations to creep up.
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Joshua Colp [Thu, 1 Feb 2007 21:54:28 +0000 (21:54 +0000)]
Merged revisions 53095 via svnmerge from
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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines
Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113)
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Russell Bryant [Thu, 1 Feb 2007 21:24:52 +0000 (21:24 +0000)]
Fix the FIELDQTY function to not crash. (reported by blitzrage and Corydon on IRC)
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Olle Johansson [Thu, 1 Feb 2007 21:15:50 +0000 (21:15 +0000)]
Going backwards, blame file.
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Joshua Colp [Thu, 1 Feb 2007 21:11:28 +0000 (21:11 +0000)]
Merged revisions 53084 via svnmerge from
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r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2 lines
Return previous behavior of having MOH pick up where it was left off. (issue #8672 reported by sinistermidget)
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Joshua Colp [Thu, 1 Feb 2007 21:06:02 +0000 (21:06 +0000)]
Make func_strings build under dev mode. Didn't I do this today already in the berkeley DB?
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Olle Johansson [Thu, 1 Feb 2007 21:05:34 +0000 (21:05 +0000)]
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
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Olle Johansson [Thu, 1 Feb 2007 20:38:58 +0000 (20:38 +0000)]
Change debug level for state change message that is not really informative when debugging app_queue
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Olle Johansson [Thu, 1 Feb 2007 20:28:54 +0000 (20:28 +0000)]
Cleaning up the devicestate callback function
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Tilghman Lesher [Thu, 1 Feb 2007 20:13:40 +0000 (20:13 +0000)]
Oops.
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Tilghman Lesher [Thu, 1 Feb 2007 20:09:52 +0000 (20:09 +0000)]
Merged revisions 53074 via svnmerge from
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r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) | 2 lines
Bug 8965 - Allow FIELDQTY to work with both variables and dialplan functions
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Joshua Colp [Thu, 1 Feb 2007 19:33:33 +0000 (19:33 +0000)]
Add missing 'F' letter to getopt so it magically becomes a valid option. (issue #8960 reported by tzafrir)
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Tilghman Lesher [Thu, 1 Feb 2007 19:21:20 +0000 (19:21 +0000)]
Merged revisions 53069 via svnmerge from
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r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) | 2 lines
No wonder FIELDQTY doesn't work with functions... the documentation in pbx.c was wrong
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