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git.ipfire.org Git - thirdparty/asterisk.git/log
Kevin P. Fleming [Fri, 6 Oct 2006 21:07:44 +0000 (21:07 +0000)]
various cleanups
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Joshua Colp [Fri, 6 Oct 2006 18:46:28 +0000 (18:46 +0000)]
When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106 reported by bungalow)
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Joshua Colp [Fri, 6 Oct 2006 16:53:48 +0000 (16:53 +0000)]
Merged revisions 44580 via svnmerge from
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r44580 | file | 2006-10-06 12:52:14 -0400 (Fri, 06 Oct 2006) | 2 lines
Even more frames to treat as though the remote side disappeared (issue #8097 reported by eldadran)
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Luigi Rizzo [Fri, 6 Oct 2006 15:59:23 +0000 (15:59 +0000)]
make sure sockets are blocking when they should be blocking.
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Christian Richter [Fri, 6 Oct 2006 12:53:41 +0000 (12:53 +0000)]
Merged revisions 44460 via svnmerge from
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r44460 | crichter | 2006-10-05 12:02:38 +0200 (Do, 05 Okt 2006) | 1 line
fixed segfault which happens during hold/transfer action
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Christian Richter [Fri, 6 Oct 2006 12:52:01 +0000 (12:52 +0000)]
Merged revisions 44335 via svnmerge from
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r44335 | crichter | 2006-10-04 17:26:59 +0200 (Mi, 04 Okt 2006) | 1 line
if INFORMATION Message come with keypad instead of called party number, we just use the keypad as called party number.
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Christian Richter [Fri, 6 Oct 2006 12:50:25 +0000 (12:50 +0000)]
Merged revisions 44334 via svnmerge from
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r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line
added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible
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Christian Richter [Fri, 6 Oct 2006 10:44:34 +0000 (10:44 +0000)]
Merged revisions 44149 via svnmerge from
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r44149 | crichter | 2006-10-02 15:28:14 +0200 (Mo, 02 Okt 2006) | 1 line
fixed the hold/retrieve/transfer issues, removed a useless bc field, added setting of frame.delivery fields, some minor code cleanups
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Joshua Colp [Thu, 5 Oct 2006 19:57:16 +0000 (19:57 +0000)]
Merged revisions 44501 via svnmerge from
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r44501 | file | 2006-10-05 15:55:41 -0400 (Thu, 05 Oct 2006) | 2 lines
Treat busy control frames as hangup in the file streaming core (issue #8097 reported by eldadran)
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Steve Murphy [Thu, 5 Oct 2006 18:21:39 +0000 (18:21 +0000)]
This mod fixes a problem pointed out by dgarstang. Many thanks to Doug\!
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Joshua Colp [Thu, 5 Oct 2006 18:01:51 +0000 (18:01 +0000)]
One more T.38 fix! Don't leave a reinvite hanging by a thread if the other side is already setup with T.38
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Kevin P. Fleming [Thu, 5 Oct 2006 16:10:01 +0000 (16:10 +0000)]
don't segfault when an argument without a close parenthesis is found
stop parsing as soon as that situation occurs
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Steve Murphy [Thu, 5 Oct 2006 15:22:37 +0000 (15:22 +0000)]
I put the accumulated changes from the commit logs and inspection, into CHANGES. Hope everyone approves\!
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Steve Murphy [Thu, 5 Oct 2006 15:04:22 +0000 (15:04 +0000)]
Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right\?
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Joshua Colp [Thu, 5 Oct 2006 02:40:40 +0000 (02:40 +0000)]
Don't totally bail out if T.38 was negotiated
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Kevin P. Fleming [Thu, 5 Oct 2006 01:42:06 +0000 (01:42 +0000)]
this change was correct, the old version is no longer needed
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Kevin P. Fleming [Thu, 5 Oct 2006 01:30:05 +0000 (01:30 +0000)]
Merged revisions 44432 via svnmerge from
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r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04 Oct 2006) | 2 lines
fix Polycom presence notification again
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Luigi Rizzo [Wed, 4 Oct 2006 22:52:11 +0000 (22:52 +0000)]
as far as i can tell astman only uses newt...
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Luigi Rizzo [Wed, 4 Oct 2006 22:50:48 +0000 (22:50 +0000)]
put linker flags in ASTLDFLAGS where they belong
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Kevin P. Fleming [Wed, 4 Oct 2006 21:17:30 +0000 (21:17 +0000)]
Merged revisions 44392 via svnmerge from
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r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04 Oct 2006) | 3 lines
remove workaround for old Polycom firmware SUBSCRIBE requests
add workaround for new Polycom firmware SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)
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Kevin P. Fleming [Wed, 4 Oct 2006 21:04:21 +0000 (21:04 +0000)]
make LOW_MEMORY builds actually work
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Steve Murphy [Wed, 4 Oct 2006 19:57:08 +0000 (19:57 +0000)]
These changes fix the problems reported in bug 8090
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Kevin P. Fleming [Wed, 4 Oct 2006 19:47:22 +0000 (19:47 +0000)]
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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Steve Murphy [Wed, 4 Oct 2006 17:04:21 +0000 (17:04 +0000)]
I've been meaning to add some explanation about muted... here it is
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Steve Murphy [Wed, 4 Oct 2006 16:49:47 +0000 (16:49 +0000)]
CLI reverbification update to this config file
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Steve Murphy [Wed, 4 Oct 2006 15:52:12 +0000 (15:52 +0000)]
OOOps! There is no alternative to suggest in 1.4. A small erasure.
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Steve Murphy [Wed, 4 Oct 2006 15:49:20 +0000 (15:49 +0000)]
In response to bug 7776, a Warning has been added to the doc string for Macro().
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Kevin P. Fleming [Wed, 4 Oct 2006 00:25:44 +0000 (00:25 +0000)]
ensure that local include files are always used
avoid a duplicate function name (term_init())
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Matt O'Gorman [Tue, 3 Oct 2006 22:35:43 +0000 (22:35 +0000)]
fix issue with dialing client without resource.
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Kevin P. Fleming [Tue, 3 Oct 2006 20:18:29 +0000 (20:18 +0000)]
Merged revisions 44296 via svnmerge from
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r44296 | kpfleming | 2006-10-03 15:14:13 -0500 (Tue, 03 Oct 2006) | 2 lines
fix a logic error in my previous fix to the queue reload code
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Paul Cadach [Tue, 3 Oct 2006 18:42:20 +0000 (18:42 +0000)]
Change default presentation indicator to "user provided not screened" if octet 3a missed in CallingPartyNumber IE
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Joshua Colp [Tue, 3 Oct 2006 18:35:55 +0000 (18:35 +0000)]
Use VideoSupport instead so it is considered a valid XML attribute name. (issue #8075 reported by renemendoza)
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Paul Cadach [Tue, 3 Oct 2006 18:30:48 +0000 (18:30 +0000)]
Fix preparation of type and presentation of calling number
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Matt O'Gorman [Tue, 3 Oct 2006 00:01:34 +0000 (00:01 +0000)]
updated res_jabber for even better component support, soon will be jep-0100 compliant.
also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version.
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Joshua Colp [Mon, 2 Oct 2006 20:11:02 +0000 (20:11 +0000)]
Merged revisions 44213 via svnmerge from
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r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2 lines
Change the fd on the I/O context in case it changed during the reload, which is indeed possible. (issue #7943 reported by eclubb)
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Joshua Colp [Mon, 2 Oct 2006 19:41:39 +0000 (19:41 +0000)]
Merged revisions 44198 via svnmerge from
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r44198 | file | 2006-10-02 15:39:59 -0400 (Mon, 02 Oct 2006) | 2 lines
We should be using $AST_SBIN instead of hardcoding the path for the error message (issue #7942 reported by eclubb)
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Paul Cadach [Mon, 2 Oct 2006 18:52:56 +0000 (18:52 +0000)]
Missed part of userconf functionality for chan_h323
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Joshua Colp [Mon, 2 Oct 2006 17:25:13 +0000 (17:25 +0000)]
Merged revisions 44168 via svnmerge from
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r44168 | file | 2006-10-02 13:22:27 -0400 (Mon, 02 Oct 2006) | 2 lines
Shrink when current_ioc is unused. It is set to -1 when unused, not 0. (issue #7941 reported by eclubb)
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Paul Cadach [Mon, 2 Oct 2006 17:16:37 +0000 (17:16 +0000)]
Typo fix
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Paul Cadach [Mon, 2 Oct 2006 17:15:11 +0000 (17:15 +0000)]
Optimization of oh323_indicate(): less locks - less problems, plus single exit point
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Mark Spencer [Mon, 2 Oct 2006 02:38:28 +0000 (02:38 +0000)]
Don't use Channel when you're not talking about a channel :)
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Paul Cadach [Sun, 1 Oct 2006 19:32:24 +0000 (19:32 +0000)]
Do not simulate any audio tones if we got PROGRESS message
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Russell Bryant [Sun, 1 Oct 2006 18:30:06 +0000 (18:30 +0000)]
Fix a problem that cuased AST_DATA_DIR in defaults.h to be empty. The cause is
that since ASTDATADIR is explicitly exported using "export ASTDATADIR" at the
top of the Makefile, make no longer considers the variable "undefined", so the
Makefile can't use ?= to set ASTDATADIR if not yet set.
(issue #8063, reported by akohlsmith, fixed by me)
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Russell Bryant [Sun, 1 Oct 2006 15:20:12 +0000 (15:20 +0000)]
Merged revisions 44110 via svnmerge from
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r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 Oct 2006) | 3 lines
Fix the name of the "eventmemberstatus" option in the sample queues.conf
(issue #8065, adamg)
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Luigi Rizzo [Sun, 1 Oct 2006 15:01:28 +0000 (15:01 +0000)]
sync with trunk - move variable declarations to the beginning of a block.
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Paul Cadach [Sat, 30 Sep 2006 19:20:38 +0000 (19:20 +0000)]
Allow one-way RTP streams (device->Asterisk)
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Luigi Rizzo [Sat, 30 Sep 2006 16:28:39 +0000 (16:28 +0000)]
fix two recent build problems:
- with AST_DEVMODE, building codecs/lpc10 fails because of lots
of warnings, and the configure step in editline fails as well.
Fix this by removing the -Werror in these steps.
- on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile
(a possible alternative way is to add "export AST_LIBS" near
the beginning of the file).
With this fix, i believe that some of the platform-specific
conditionals in main/Makefile are redundant (because they should
be already dealt with in the top level Makefile) but i don't
have a platform to check.
Merging to head will happen in a moment.
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Paul Cadach [Sat, 30 Sep 2006 16:12:23 +0000 (16:12 +0000)]
Fix issue #7928 correctly. Next is a comment of previous fix:
Issue #7928 - Don't send both 404 and 503. Fix by phsultan with
a small fix by me, myself or I. Thanks, Philippe!
(This was caused by my changes to the transaction handling)
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Paul Cadach [Sat, 30 Sep 2006 04:37:39 +0000 (04:37 +0000)]
Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.
If someone else knows how to provide workaround for such cases,
please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
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Kevin P. Fleming [Fri, 29 Sep 2006 22:51:53 +0000 (22:51 +0000)]
ignore temporary files made by the Makefiles during a build
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Kevin P. Fleming [Fri, 29 Sep 2006 22:47:40 +0000 (22:47 +0000)]
fix a few build system bugs, and convert Makefiles to be compatible with GNU make 3.80
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Jason Parker [Fri, 29 Sep 2006 22:35:09 +0000 (22:35 +0000)]
Fix a bug with the removal of 'atleast' argument to 'core verbose' and 'core debug'.
Add that argument back in.
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Paul Cadach [Fri, 29 Sep 2006 21:09:10 +0000 (21:09 +0000)]
Compile first, please
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Paul Cadach [Fri, 29 Sep 2006 21:05:43 +0000 (21:05 +0000)]
Set TON/PRESENTATION information more carefully when no CallingNumber IE available
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Paul Cadach [Fri, 29 Sep 2006 20:43:13 +0000 (20:43 +0000)]
Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented)
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Paul Cadach [Fri, 29 Sep 2006 19:09:22 +0000 (19:09 +0000)]
Ported code refers to H.450 - add includes
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Paul Cadach [Fri, 29 Sep 2006 19:06:55 +0000 (19:06 +0000)]
Properly pass TON/PRESENTATION information - original
H323Connection::SendSignalSetup() destroys Q.931 fields.
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Kevin P. Fleming [Fri, 29 Sep 2006 18:49:07 +0000 (18:49 +0000)]
yet another place where we were not using the correct CFLAGS by default
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Kevin P. Fleming [Fri, 29 Sep 2006 18:40:17 +0000 (18:40 +0000)]
missed one conversion to ASTCFLAGS
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Paul Cadach [Fri, 29 Sep 2006 18:30:34 +0000 (18:30 +0000)]
Pass TON/PRESENTATION information too
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Kevin P. Fleming [Fri, 29 Sep 2006 18:25:49 +0000 (18:25 +0000)]
don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44008
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Kevin P. Fleming [Fri, 29 Sep 2006 16:52:27 +0000 (16:52 +0000)]
support --without-curl in configure script
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43997
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Kevin P. Fleming [Fri, 29 Sep 2006 16:47:05 +0000 (16:47 +0000)]
another cross-compile fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43996
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Kevin P. Fleming [Fri, 29 Sep 2006 16:38:27 +0000 (16:38 +0000)]
a couple more environment settings that can't leak into the menuselect build
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43993
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Kevin P. Fleming [Fri, 29 Sep 2006 13:45:40 +0000 (13:45 +0000)]
proper fix for ast_group_t change
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43978
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Kevin P. Fleming [Thu, 28 Sep 2006 22:42:18 +0000 (22:42 +0000)]
eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43952
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Jason Parker [Thu, 28 Sep 2006 20:11:22 +0000 (20:11 +0000)]
Fix incorrect argument order for member names, on persisted members.
Issue 8047, patch by jmls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43944
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Joshua Colp [Thu, 28 Sep 2006 18:05:43 +0000 (18:05 +0000)]
Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43933
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Joshua Colp [Thu, 28 Sep 2006 18:01:58 +0000 (18:01 +0000)]
Blocked revisions 43924 via svnmerge
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r43924 | file | 2006-09-28 14:00:30 -0400 (Thu, 28 Sep 2006) | 2 lines
Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43932
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Kevin P. Fleming [Thu, 28 Sep 2006 17:35:42 +0000 (17:35 +0000)]
Merged revisions 43916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28 Sep 2006) | 2 lines
fix buggy (and overly complex) loop used during reload of app_queue for static member list updating
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43919
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Paul Cadach [Thu, 28 Sep 2006 17:34:19 +0000 (17:34 +0000)]
Extend call establishment timeout
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43918
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Joshua Colp [Thu, 28 Sep 2006 17:31:09 +0000 (17:31 +0000)]
Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43915
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Joshua Colp [Thu, 28 Sep 2006 17:14:07 +0000 (17:14 +0000)]
Warning be gone!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43913
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BJ Weschke [Thu, 28 Sep 2006 16:41:05 +0000 (16:41 +0000)]
Merged revisions 43897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 Sep 2006) | 3 lines
app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach)
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Kevin P. Fleming [Thu, 28 Sep 2006 16:38:25 +0000 (16:38 +0000)]
Merged revisions 43895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28 Sep 2006) | 2 lines
eliminate compiler warning introduced by recent changes
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Joshua Colp [Thu, 28 Sep 2006 16:17:36 +0000 (16:17 +0000)]
Merged revisions 43891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2 lines
Stop the stream after waitstream returns so that our formats get restored. (issue #7370 reported by kryptolus)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43893
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Paul Cadach [Thu, 28 Sep 2006 15:56:21 +0000 (15:56 +0000)]
Fix compiler warning
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43877
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BJ Weschke [Thu, 28 Sep 2006 15:29:21 +0000 (15:29 +0000)]
Merged revisions 43871 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28 Sep 2006) | 3 lines
Fix race condion crash with get_member_status (#7864 - tim_ringenbach reported and patched)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43873
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BJ Weschke [Thu, 28 Sep 2006 13:24:10 +0000 (13:24 +0000)]
Autopause not working for queue members. (#8042 - jmls reported and patch)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43864
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Paul Cadach [Thu, 28 Sep 2006 12:58:22 +0000 (12:58 +0000)]
Force remote side to start media on outgoing PROGRESS message
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43862
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Paul Cadach [Thu, 28 Sep 2006 12:47:23 +0000 (12:47 +0000)]
Put attribute tag at correct place
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Christian Richter [Thu, 28 Sep 2006 11:03:05 +0000 (11:03 +0000)]
Merged revisions 43764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 Sep 2006) | 1 line
fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed.
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Paul Cadach [Thu, 28 Sep 2006 10:51:21 +0000 (10:51 +0000)]
Do not open transmit channel until TCS is received
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43846
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Paul Cadach [Thu, 28 Sep 2006 06:02:45 +0000 (06:02 +0000)]
Don't warn on HOLD/UNHOLD control frames
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43844
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Paul Cadach [Thu, 28 Sep 2006 06:01:37 +0000 (06:01 +0000)]
Don't treat unknown control frames as voice
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43843
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Tilghman Lesher [Wed, 27 Sep 2006 20:21:54 +0000 (20:21 +0000)]
Merged revisions 43815 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27 Sep 2006) | 2 lines
Avoid inability to lock directory log message by creating the directory ahead of time. (Issue 7631)
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Jason Parker [Wed, 27 Sep 2006 19:44:02 +0000 (19:44 +0000)]
Fix an issue with PLAYBACKSTATUS not being set under certain circumstances.
Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string.
Fix Background() to return -1 like Playback(), if no args are specified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43803
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Jason Parker [Wed, 27 Sep 2006 19:36:23 +0000 (19:36 +0000)]
Blocking this from 1.4, fix is quite a bit different.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43801
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Joshua Colp [Wed, 27 Sep 2006 19:10:59 +0000 (19:10 +0000)]
Compensate for out of order packets better if RFC2833 compensation is turned on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43798
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Joshua Colp [Wed, 27 Sep 2006 17:00:31 +0000 (17:00 +0000)]
Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43783
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Russell Bryant [Wed, 27 Sep 2006 16:55:49 +0000 (16:55 +0000)]
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43779
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Christian Richter [Wed, 27 Sep 2006 16:24:51 +0000 (16:24 +0000)]
removed the chan_misdn versioning, since asterisk has it's own
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43775
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Joshua Colp [Wed, 27 Sep 2006 16:23:12 +0000 (16:23 +0000)]
Make rfc2833compensate a global option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43774
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Russell Bryant [Wed, 27 Sep 2006 04:35:18 +0000 (04:35 +0000)]
Backport revision 43754 from the trunk, which removes an unused buffer from
mm_login to close bug 8038, as well as addresses some formatting and coding
guidelines issues in passing.
Originally, I did not commit this to 1.4 since it is not necessarily fixing a
bug. However, since the IMAP storage code is brand new, I decided it would
be better to make the change here as well, in case someone has to work on this
code to address issues in the very near future. I don't want to make
unnecessary merge problems going to the trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43756
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Steve Murphy [Wed, 27 Sep 2006 02:32:47 +0000 (02:32 +0000)]
This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43739
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Russell Bryant [Tue, 26 Sep 2006 20:56:42 +0000 (20:56 +0000)]
(This was actually BE-65)
Merged revisions 43708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | 7 lines
Back in revision 4798, this message was changed from using ast_cli() to directly
calling write(). During this change, checking if this was a remote console was
removed. This caused this message about using "exit" or "quit" to exit an
Asterisk console to come up in times where it did not make sense. This change
restores the check to see if this is a remote console before printing the
message. (fixes BE-4)
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Joshua Colp [Tue, 26 Sep 2006 20:47:26 +0000 (20:47 +0000)]
Merged revisions 43705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2 lines
Use proper type to represent the group variable (issue #8025 reported by makoto)
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Russell Bryant [Tue, 26 Sep 2006 20:30:36 +0000 (20:30 +0000)]
Add missing newline character in the warning message about deprecated TOS values
in configuration.
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Russell Bryant [Tue, 26 Sep 2006 20:24:39 +0000 (20:24 +0000)]
Merged revisions 43699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26 Sep 2006) | 6 lines
When parsing the sections of voicemail.conf that contain mailbox definitions,
don't introduce a length limit on the definition by using a 256 byte temporary
storage buffer. Instead, make the temporary buffer just as big as it needs
to be to hold the entire mailbox definition.
(fixes BE-68)
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Joshua Colp [Tue, 26 Sep 2006 20:19:33 +0000 (20:19 +0000)]
Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43697
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