Philippe Sultan
4c8559e441
Merged revisions 175089 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines
Issue a warning message if our candidate's IP is the loopback address.
(closes issue #13985 )
Reported by: jcovert
Tested by: phsultan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 14:27:59 +00:00
Philippe Sultan
e28edbe89c
Merged revisions 175058 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines
Merged revisions 175029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines
Set the initiator attribute to lowercase in our replies when receiving calls.
This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.
(closes issue #13984 )
Reported by: jcovert
Patches:
chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert
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2009-02-12 10:42:03 +00:00
Philippe Sultan
9bd17e6a65
Merged revisions 139281 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) | 5 lines
Fix two memory leaks in chan_gtalk, thanks Eliel!
(closes issue #13310 )
Reported by: eliel
Patches:
chan_gtalk.c.patch uploaded by eliel (license 64)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-21 10:02:32 +00:00
Brett Bryant
559bbfc457
Merged revisions 130129 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines
Janitor patch to change uses of sizeof to ARRAY_LEN
(closes issue #13054 )
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11 18:14:27 +00:00
Philippe Sultan
029ac4cb4d
Merged revisions 119741 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085 )
Reported by: junky
Tested by: phsultan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-02 14:40:42 +00:00
Philippe Sultan
01f7b2b579
Merged revisions 118020 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines
- remove whitespaces between tags in received XML packets before giving
them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647 )
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@118021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-23 11:02:21 +00:00
Tilghman Lesher
4ab60760e3
Merged revisions 114709 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines
Merged revisions 114708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534 )
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2008-04-28 04:54:13 +00:00
Philippe Sultan
8205252c82
Merged revisions 112821 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r112821 | phsultan | 2008-04-04 21:28:49 +0200 (Fri, 04 Apr 2008) | 9 lines
Merged revisions 112820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line
Free newly allocated channel before returning
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2008-04-04 19:30:35 +00:00
Philippe Sultan
acf6ac5a7f
Merged revisions 112785 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines
Merged revisions 112766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines
Prevent call connections when codecs don't match.
(closes issue #10604 )
Reported by: keepitcool
Patches:
branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
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2008-04-04 17:43:01 +00:00
Jason Parker
2b1b649a97
Merged revisions 107718 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/trunk
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r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | 13 lines
Merged revisions 107714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines
Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber).
(closes issue #12014 )
Reported by: junky
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2008-03-11 20:54:38 +00:00
Philippe Sultan
7293986e44
Remove unnecessary if statements before calling iks_delete (redundant check is
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done inside iks_delete), thus making the code conform with coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-29 14:15:03 +00:00
Philippe Sultan
55240a4e35
Merged revisions 97489 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines
Set the caller id within the gtalk_alloc function.
As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549 .
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2008-01-09 16:59:09 +00:00
Olle Johansson
d2b29df4f0
Manager events from the "moremanager" branch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27 19:50:12 +00:00
Luigi Rizzo
7e8835e0d7
remove another set of redundant #include "asterisk/options.h"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:24:55 +00:00
Luigi Rizzo
0595b5e2aa
include "logger.h" and errno.h from asterisk.h - usage shows that they
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were included almost everywhere.
Remove some of the instances.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 18:52:04 +00:00
Luigi Rizzo
d82a631f9c
more removal of duplicate #include lines
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17 00:02:33 +00:00
Luigi Rizzo
fdb7f7ba3d
Start untangling header inclusion in a way that does not affect
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Tilghman Lesher
7c56918262
Commit some cleanups to the format type code.
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06 22:51:48 +00:00
Jason Parker
2c582c7cfb
Allow gtalk and jingle to use TLS connections again.
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Closes issue #9972
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2007-11-06 18:44:19 +00:00
Jason Parker
2902601eea
Remove traces of gnutls, since we no longer use/need it.
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2007-11-01 23:26:51 +00:00
Jason Parker
fa33494d80
Merged revisions 87906 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11130 )
(closes issue #11132 )
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r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines
Don't try to allocate memory that we're just going to re-allocate later anyways.
Issues 11130 and 11132, patch by eliel.
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2007-10-31 21:18:52 +00:00
Jason Parker
ebe4050128
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 20:05:18 +00:00
Jason Parker
b0f3e6097e
Convert NEW_CLI to AST_CLI.
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Closes issue #11039 , as suggested by seanbright.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19 18:29:40 +00:00
Philippe Sultan
65547b09b4
Fix CLI help output
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16 10:38:57 +00:00
Russell Bryant
e97a723cf1
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
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(closes issue #10724 )
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
Philippe Sultan
5734c0df49
Merged revisions 82309 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 Sep 2007) | 4 lines
Closes issue #9401 , reported and patched by irrot, with slight
modifications by me.
Handle DTMF sent by Asterisk properly.
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2007-09-13 11:54:56 +00:00
Philippe Sultan
da620112de
Merged revisions 81743 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) | 1 line
Various string length fixes. Removed an unused variable in aji_client structure (context)
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2007-09-06 17:00:58 +00:00
Philippe Sultan
2fd2667d13
Merged revisions 81410 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 Aug 2007) | 3 lines
Make the 'gtalk show channels' CLI command available.
Closes issue 10548, reported by keepitcool.
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2007-08-31 17:43:50 +00:00
Philippe Sultan
fce77d7d6a
Merged revisions 80661 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines
Closes issue #10509
Googletalk calls are answered too early, which results in CDRs wrongly
stating that a call was ANSWERED when the calling party cancelled a
call before before being established.
We must not answer the call upon reception of a 'transport-accept' iq
packet, but this packet still needs to be acknowledged, otherwise the
remote peer would close the call (like in #8970 ).
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2007-08-24 11:49:36 +00:00
Tilghman Lesher
56b9568164
Don't reload a configuration file if nothing has changed.
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2007-08-16 21:09:46 +00:00
Joshua Colp
d5eda8709c
Merged revisions 79174 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437 )
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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2007-08-13 14:22:46 +00:00
Joshua Colp
22114b509d
Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
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2007-08-08 21:44:58 +00:00
Joshua Colp
4897378ff1
Merged revisions 72331 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines
Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot)
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2007-06-27 22:58:53 +00:00
Jason Parker
8bf745194f
Merged revisions 72125 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines
Don't modify a variable that we don't want modified. Make a copy of it instead.
Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).
Note: chan_jingle in trunk does not appear to have the same bug.
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2007-06-27 17:14:31 +00:00
Russell Bryant
3957ce9215
Merged revisions 70084 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines
Only attempt to queue a hangup on the owner channel if it actually exists.
(issue #9795 , patch from zandbelt)
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2007-06-19 19:15:03 +00:00
Russell Bryant
055d82cbce
Add a massive set of changes for converting to use the ast_debug() macro.
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(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14 19:39:12 +00:00
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
Tilghman Lesher
ce9ec91897
ast_calloc janitor (Inspired by issue 9860)
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2007-06-03 06:10:27 +00:00
Kevin P. Fleming
c74518e3ff
Merged revisions 66157 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines
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
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2007-05-25 14:37:55 +00:00
Kevin P. Fleming
287bcc6127
Merged revisions 65965-65967 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines
don't use uninitialized variables
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r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines
don't reference GnuTLS headers and functions unless the configure script found it
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r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines
oops, use #ifdef instead of #if
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2007-05-24 19:05:42 +00:00
Olle Johansson
b4f7d35240
Merged revisions 65901 via svnmerge from
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r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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2007-05-24 15:29:10 +00:00
Olle Johansson
bd2ae8b587
Merged revisions 65892 via svnmerge from
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r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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2007-05-24 15:23:04 +00:00
Olle Johansson
45bb9430b9
Merged revisions 65857 via svnmerge from
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r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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2007-05-24 15:21:39 +00:00
Olle Johansson
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Merged revisions 65841 via svnmerge from
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r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines
Issue #8536 - Caller ID not set in CDR for jingle
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2007-05-24 14:52:01 +00:00
Steve Murphy
ecaf781933
Merged revisions 60989 via svnmerge from
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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2007-04-10 05:41:34 +00:00
Russell Bryant
b2ddaaf033
Add support for RTP packetization in chan_jingle and chan_gtalk.
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(issue #9416 , phsultan)
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2007-04-03 22:33:03 +00:00
Jason Parker
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r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines
Fix locking issue, and accept "transport-accept" as a valid accept message.
This should solve issues 8970 and 8503.
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2007-02-21 20:30:54 +00:00
Jason Parker
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Merged revisions 55799 via svnmerge from
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r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines
Fix segfault when buddy couldn't be found.
Issue 7764, patch by sailer
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2007-02-21 02:04:10 +00:00
Jason Parker
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r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines
No need to cast nor free with strdupa (thanks file)
55555!
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2007-02-20 16:56:58 +00:00
Olle Johansson
ba32ee49d0
Adding Realtime Text support (T.140) to Asterisk
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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2007-02-16 13:35:44 +00:00