Russell Bryant
0dc5766279
Add the new options for attended transfer to the CHANGES file.
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2007-05-02 15:20:41 +00:00
Russell Bryant
65ac2341c0
For some reason when I merged 802.1p support, the new documentation file was
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not properly added. Thanks to IgorG for pointing it out! :)
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2007-05-02 15:17:26 +00:00
Olle Johansson
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
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2007-05-02 12:12:02 +00:00
Olle Johansson
1d51b2e161
More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
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But first, some serious SIP testing :-)
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2007-05-02 12:00:03 +00:00
Olle Johansson
8fee67c83b
Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
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2007-05-02 09:41:03 +00:00
Olle Johansson
daefa6a8b4
Merged revisions 62624 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines
Don't unlock a channel that we already know does not exist (propably isue 8228)
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2007-05-02 09:35:14 +00:00
Olle Johansson
2a8a2f18b3
Updating CREDITS
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2007-05-02 05:57:53 +00:00
Russell Bryant
7df63e233c
In addition to making it so attended transfers don't fail unnecessarily,
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add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413 , patch from sergee with very minor modifications by me)
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2007-05-01 22:24:51 +00:00
Russell Bryant
37d94ceab9
Merged revisions 62548 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines
Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines
Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338 , patch by svanlund)
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2007-05-01 21:57:58 +00:00
Tilghman Lesher
2059f767dd
Merged revisions 62545 via svnmerge from
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r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007) | 2 lines
Bug 9590 - Memory leaks around find_user() (found by rayjay, different fixes by me)
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2007-05-01 21:41:48 +00:00
Russell Bryant
32ca8db474
Merged revisions 62497 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62497 | russell | 2007-05-01 11:26:48 -0500 (Tue, 01 May 2007) | 11 lines
Merged revisions 62496 via svnmerge from
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r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines
Add indications.conf information for the Philippines.
(issue #9525 , reported and patched by loloski)
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2007-05-01 16:27:14 +00:00
Russell Bryant
c82fd9020f
Add a note to CHANGES about the new support for 802.1p. Thanks IgorG!
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2007-04-30 21:54:13 +00:00
Russell Bryant
683417407e
This patch adds additional information to the EXITWITHKEY and EXITWITHTIMEOUT
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entries in the queue log.
(issue #7561 , reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
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2007-04-30 16:46:49 +00:00
Russell Bryant
b419fc1134
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540 , patch submitted by IgorG)
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2007-04-30 16:16:26 +00:00
Russell Bryant
a91f9b138d
Merged revisions 62419 via svnmerge from
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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines
Merged revisions 62417 via svnmerge from
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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2007-04-30 15:59:27 +00:00
Russell Bryant
5cb08adc7a
Don't crash when invalid arguments are provided to the CHANNEL() function
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for a SIP channel.
(issue #9619 , reported by jtodd, original patch by Corydon76, committed patch
slightly modified by me)
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2007-04-30 15:37:23 +00:00
Russell Bryant
3181a91148
Merged revisions 62414 via svnmerge from
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r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | 4 lines
When serving dynamic content, include a Cache-Control header to instruct the
browsers to not store the resulting content.
(issue #9621 , reported by Pari, patch by me)
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2007-04-30 15:30:02 +00:00
Jason Parker
b942fe3d89
Merged revisions 62371 via svnmerge from
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r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr 2007) | 2 lines
Remove unused (and potentially confusing) jitterbuffer options from sample config.
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2007-04-30 14:56:43 +00:00
Joshua Colp
fe158ded8a
Merged revisions 62369 via svnmerge from
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r62369 | file | 2007-04-30 11:36:11 -0300 (Mon, 30 Apr 2007) | 10 lines
Merged revisions 62368 via svnmerge from
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r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 lines
Update copyright notice. It's now the year 2007!
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2007-04-30 14:37:47 +00:00
Russell Bryant
0efe511879
Merged revisions 62331 via svnmerge from
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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines
Fix a bug that made the "language" setting in zapata.conf not
functional. (issue #9626 , reported and fixed by sergee)
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2007-04-29 05:51:18 +00:00
Russell Bryant
28244e7462
Blocked revisions 62299 via svnmerge
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r62299 | russell | 2007-04-28 16:56:20 -0500 (Sat, 28 Apr 2007) | 2 lines
Note that the "talker optimization" option will be enabled by default in 1.6
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2007-04-28 21:56:46 +00:00
Russell Bryant
a4a2e973ec
note MeetMe change in CHANGES
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2007-04-28 21:55:00 +00:00
Russell Bryant
dd601f4f38
Enable the functionality of the 'o' option to "optimize talker" by default.
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2007-04-28 21:54:44 +00:00
Russell Bryant
7d2102c081
Reformat some of iax2.h and convert comments to doxygen format
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2007-04-28 21:26:00 +00:00
Russell Bryant
b6b1bf3213
Merge changes from team/russell/events
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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2007-04-28 21:01:44 +00:00
Russell Bryant
503b75f885
Update the DUNDi section of the documentation with example usage of DUNDIQUERY
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and DUNDIRESULT. Also, update the automatically generated application docs.
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2007-04-28 19:53:12 +00:00
Russell Bryant
c59b8876aa
Merge changes from team/russell/dundi_results
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This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.
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2007-04-28 19:52:37 +00:00
Russell Bryant
3f414d2aea
Remove a message that goes to LOG_ERROR that's not really an error.
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2007-04-28 19:23:46 +00:00
Russell Bryant
672fbc1f81
Add a min-announce-frequency option to queues.conf which allows you to control the
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minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604 , patch by Matthew Roth)
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2007-04-27 22:08:54 +00:00
Russell Bryant
c2468b4c32
Merged revisions 62218 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615 , #9616 , reported by jiddings and BigJimmy, and fixed by me)
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2007-04-27 21:11:46 +00:00
Jason Parker
fdbb94bb19
Merged revisions 62174 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, 27 Apr 2007) | 11 lines
Merged revisions 62173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines
This transcoder message needn't be a NOTICE.
I've seen it cause confusion more than a few times.
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2007-04-27 16:18:51 +00:00
Russell Bryant
ec0cae93a4
Merged revisions 62171 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613 , reported by callguy, fixed by me)
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2007-04-27 16:15:47 +00:00
Olle Johansson
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
Olle Johansson
f9c592e50c
Merged revisions 62137 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
Merged revisions 62126 via svnmerge from
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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2007-04-27 14:37:10 +00:00
Olle Johansson
249ecdec64
Blocking patch to 1.4 that was alredy in trunk
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2007-04-27 08:22:41 +00:00
Joshua Colp
14d8979b2f
Merged revisions 62038 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines
Merged revisions 62037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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2007-04-26 16:35:14 +00:00
Russell Bryant
007fa5e0be
Merged revisions 62005 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines
Missed an ast_app_group_discard during merge. Thanks blitzrage!
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2007-04-26 03:24:01 +00:00
Joshua Colp
a212ff5a78
Merged revisions 61961 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2 lines
Don't always say that the channel is being paused if it is actually being unpaused in the Manager ack message. (reported by jsmith in #asterisk-bugs)
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2007-04-26 01:50:02 +00:00
Joshua Colp
489f9c68c3
Merged revisions 61959 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61959 | file | 2007-04-25 21:27:18 -0400 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61958 via svnmerge from
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r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 lines
Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn)
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2007-04-26 01:29:23 +00:00
Kevin P. Fleming
ee95074173
Merged revisions 61914 via svnmerge from
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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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2007-04-25 22:34:58 +00:00
Russell Bryant
dc7514a746
Merged revisions 61870 via svnmerge from
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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61866 via svnmerge from
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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2007-04-25 22:01:37 +00:00
Russell Bryant
891a005706
Merged revisions 61863 via svnmerge from
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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61862 via svnmerge from
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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2007-04-25 21:15:19 +00:00
Joshua Colp
8b2b3e172b
Merged revisions 61805 via svnmerge from
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r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines
Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-25 19:27:42 +00:00
Russell Bryant
f5d4a16cda
Merged revisions 61799 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines
Merged revisions 61798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-25 16:23:00 +00:00
Russell Bryant
a751038fbe
Merged revisions 61787 via svnmerge from
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r61787 | russell | 2007-04-24 16:34:53 -0500 (Tue, 24 Apr 2007) | 12 lines
Merged revisions 61786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines
Don't crash if a manager connection provides a username that exists in
manager.conf but does not have a password, and also requests MD5
authentication. (ASA-2007-012)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24 21:37:00 +00:00
Dwayne M. Hubbard
f3ab33014a
removed #if 0 block from chan_zap restart_monitor()
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24 19:08:28 +00:00
Russell Bryant
94459660a3
Merged revisions 61781 via svnmerge from
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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24 19:03:16 +00:00
Russell Bryant
90f152b24f
Merged revisions 61774 via svnmerge from
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r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24 16:17:36 +00:00
Joshua Colp
721f85d084
Merged revisions 61772 via svnmerge from
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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines
Merged revisions 61771 via svnmerge from
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24 16:10:10 +00:00
Russell Bryant
fa9d9af754
When building a JSON encoded string in the GetConfigJSON manager action, escape
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the '\' and '"' characters. (issue #9475 , reported by pari, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-23 18:49:19 +00:00