Tilghman Lesher [Tue, 22 Jan 2008 20:42:47 +0000 (20:42 +0000)]
Merged revisions 99643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008) | 2 lines
Fix the defines for OS X (and Solaris, too)
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Russell Bryant [Tue, 22 Jan 2008 20:41:05 +0000 (20:41 +0000)]
Make sure the command is not just present but is also configured to be executed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99645
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Olle Johansson [Tue, 22 Jan 2008 20:35:10 +0000 (20:35 +0000)]
Add a generic function to set the bridged call PVT unique id string
as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99644
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Russell Bryant [Tue, 22 Jan 2008 20:33:16 +0000 (20:33 +0000)]
Change the Asterisk CLI startup commands feature to read commands to run from cli.conf
after a discussion on the -dev list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99642
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Olle Johansson [Tue, 22 Jan 2008 17:46:43 +0000 (17:46 +0000)]
Merged revisions 99594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines
Add more dependencies on chan_local and add a note to the description of chan_local
so that people don't disable it in menuselect just to clean up.
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Olle Johansson [Tue, 22 Jan 2008 17:42:27 +0000 (17:42 +0000)]
Merged revisions 99592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 lines
Add dependency on chan_local to app_dial.
Dial still runs without chan_local, but will be missing forwarding functionality.
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Tilghman Lesher [Tue, 22 Jan 2008 17:15:08 +0000 (17:15 +0000)]
Merged revisions 99540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) | 7 lines
Ensure that we can get an address even when we don't have a default route.
(closes issue #9225)
Reported by: junky
Patches:
20080122__bug9225.diff.txt uploaded by Corydon76 (license 14)
Tested by: oej, loloski, sergee
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Russell Bryant [Tue, 22 Jan 2008 16:55:55 +0000 (16:55 +0000)]
Point out a bug in some debug counter handling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99542
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Olle Johansson [Tue, 22 Jan 2008 15:25:43 +0000 (15:25 +0000)]
Add authentication options to the SIP dialstring.
Documentation follows separately
(issue #11587)
Reported by: sobomax
Patches:
chan_sip.c-trunk.diff uploaded by sobomax (license 359)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99521
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Olle Johansson [Tue, 22 Jan 2008 09:57:16 +0000 (09:57 +0000)]
Documentation updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99483
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Olle Johansson [Tue, 22 Jan 2008 09:46:28 +0000 (09:46 +0000)]
Small fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99482
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Olle Johansson [Tue, 22 Jan 2008 08:58:46 +0000 (08:58 +0000)]
Doxygen updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99464
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Mark Michelson [Mon, 21 Jan 2008 23:56:40 +0000 (23:56 +0000)]
Merged revisions 99426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines
Fixing an issue wherein monitoring local channels was not possible. During a channel
masquerade, the monitors on the two channels involved are swapped. In 99% of the cases
this results in the desired effect. However, if monitoring a local channel, this caused
the monitor which was on the local channel to get moved onto a channel which is immediately
hung up after the masquerade has completed. By swapping the monitors prior to the masquerade,
we avoid the problem by tricking the masquerade into placing the monitor back onto the channel
where we want it.
During the investigation of the issue, the channel's monitor was the only thing that was swapped
in such a manner which did not make sense to have done. All other variable swapping made sense.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99427
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Jason Parker [Mon, 21 Jan 2008 23:25:42 +0000 (23:25 +0000)]
Fix distinctive ring detection.
Reported by: milazzo
Patches:
drings.diff uploaded by milazzo (license 383)
Closes issue #11799
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99424
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Mark Michelson [Mon, 21 Jan 2008 22:32:13 +0000 (22:32 +0000)]
Adding the QUEUENAME variable to the variables set using the setqueuevar option
in queues.conf.
Suggestion comes from Shaun2222 on IRC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99406
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Olle Johansson [Mon, 21 Jan 2008 21:11:58 +0000 (21:11 +0000)]
Remove compiler warning for uninitialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99384
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Olle Johansson [Mon, 21 Jan 2008 21:10:07 +0000 (21:10 +0000)]
Doxygen updates.
The TCP/TLS code was committed without any doxygen obviously. Tss tss.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99383
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Olle Johansson [Mon, 21 Jan 2008 20:41:15 +0000 (20:41 +0000)]
Updating doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99382
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Tilghman Lesher [Mon, 21 Jan 2008 18:15:57 +0000 (18:15 +0000)]
Merged revisions 99341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver.
(closes issue #11798)
Reported by: Corydon76
Patches:
20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak
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Joshua Colp [Mon, 21 Jan 2008 16:02:06 +0000 (16:02 +0000)]
Merged revisions 99301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 lines
Bump the buffer size for Via headers up to 512. There are some exceptionally large Via headers out there.
(closes issue #11783)
Reported by: ofirroval
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Olle Johansson [Mon, 21 Jan 2008 07:02:08 +0000 (07:02 +0000)]
Update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99280
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Joshua Colp [Mon, 21 Jan 2008 03:54:47 +0000 (03:54 +0000)]
Change over to using ast_debug so these debug messages don't always show up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99265
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Russell Bryant [Sun, 20 Jan 2008 07:28:23 +0000 (07:28 +0000)]
Add a "console active" CLI command, which lets you find out which console device
is currently active for the Asterisk CLI, or to set it. Also, knock multiple device
support off of the to-do list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99248
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Russell Bryant [Sun, 20 Jan 2008 06:13:22 +0000 (06:13 +0000)]
correct the name of a CLI command for getting available device names
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99232
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Russell Bryant [Sun, 20 Jan 2008 06:11:49 +0000 (06:11 +0000)]
Merge changes from team/russell/console_devices
- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99227
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Russell Bryant [Sat, 19 Jan 2008 10:06:02 +0000 (10:06 +0000)]
Merged revisions 99187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | 4 lines
Fix a couple of memory leaks with frame handling. Specifically,
ast_frame_free() needed to be called on the frame that came from the translator
to signed linear.
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Russell Bryant [Sat, 19 Jan 2008 05:28:33 +0000 (05:28 +0000)]
Add Cygwin as an "other" platform that is supported
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99167
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Russell Bryant [Sat, 19 Jan 2008 05:26:46 +0000 (05:26 +0000)]
Various README updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99166
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Joshua Colp [Fri, 18 Jan 2008 22:58:50 +0000 (22:58 +0000)]
Blocked revisions 99127 via svnmerge
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r99127 | file | 2008-01-18 18:57:15 -0400 (Fri, 18 Jan 2008) | 2 lines
Remove the __ in front of the unused variable. This causes some compilers to freak out.
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Russell Bryant [Fri, 18 Jan 2008 22:04:33 +0000 (22:04 +0000)]
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085
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Russell Bryant [Fri, 18 Jan 2008 21:38:01 +0000 (21:38 +0000)]
Merged revisions 99081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines
Revert adding the packed attribute, as it really doesn't make sense why that
would do any good. Fix the real bug, which is to do the check to see if the
frame came from a translator at the beginning of ast_frame_free(), instead of
at the end. This ensures that it always gets checked, even if none of the
parts of the frame are malloc'd, and also ensures that we aren't looking at
free'd memory in the case that it is a malloc'd frame.
(closes issue #11792, reported by explidous, patched by me)
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Russell Bryant [Fri, 18 Jan 2008 21:24:05 +0000 (21:24 +0000)]
Merged revisions 99079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | 4 lines
Since we're relying on the offset between the frame and the beginning of the translator
pvt struct, set the packed attribute to make sure we get to the right place.
(potential fix for issue #11792)
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Terry Wilson [Fri, 18 Jan 2008 16:58:50 +0000 (16:58 +0000)]
This should at least temporarily fix a problem where the 't' Dial
option is incorrectly passed to the transferee when built-in
attended transfers are used. There is still a problem with 'T',
but better to fix some problems than no problems while we work
on it.
(closes issue #7904)
Reported by: k-egg
Patches:
transfer-fix-trunk-r97657.diff uploaded by sergee (license 138)
Tested by: sergee, otherwiseguy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99026
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Tilghman Lesher [Fri, 18 Jan 2008 06:58:35 +0000 (06:58 +0000)]
Convert func_odbc to use SQLExecDirect for speed
(closes issue #10723)
Reported by: mnicholson
Patches:
func-odbc-direct-execute1.diff uploaded by mnicholson (license 96)
Tested by: Corydon76, mnicholson, falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99018
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Tilghman Lesher [Fri, 18 Jan 2008 06:52:18 +0000 (06:52 +0000)]
Permit username and password to be NULL (which enables pass-through from the layer above).
Reported by: lurcher
Patch by: tilghman
(Closes issue #11739)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99017
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Tilghman Lesher [Fri, 18 Jan 2008 02:06:27 +0000 (02:06 +0000)]
Reset default CUT delimiter back to '-'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99015
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Russell Bryant [Thu, 17 Jan 2008 23:28:16 +0000 (23:28 +0000)]
Make the output of "console list devices" a bit prettier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99011
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Russell Bryant [Thu, 17 Jan 2008 23:21:30 +0000 (23:21 +0000)]
List which devices are inputs and outputs in "console list devices"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99009
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Russell Bryant [Thu, 17 Jan 2008 23:20:37 +0000 (23:20 +0000)]
Add AST_FORMAT_SLINEAR16 to the list for ast_best_codec()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99008
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Russell Bryant [Thu, 17 Jan 2008 22:50:13 +0000 (22:50 +0000)]
Merged revisions 99004 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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Mark Michelson [Thu, 17 Jan 2008 22:22:02 +0000 (22:22 +0000)]
Fixing trunk IMAP build
(closes issue #11788)
Reported by: DEA
Patches:
vm-imap-build-fix.txt uploaded by DEA (license 3)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99002
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Jason Parker [Thu, 17 Jan 2008 20:51:26 +0000 (20:51 +0000)]
Add several busy detection related defines to menuselect.
Allow better busy detect debugging (with BUSYDETECT_DEBUG).
Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines.
(closes issue #11107)
Patches:
busydetect_enhancement.patch uploaded by agx (license 298)
busydetect-r94975.diff uploaded by sergee (license 138)
Additional changes/cleanup by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98998
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Mark Michelson [Thu, 17 Jan 2008 16:33:24 +0000 (16:33 +0000)]
state_interface could be NULL, so use the never-NULL cur->state_interface for this check
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98994
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Mark Michelson [Thu, 17 Jan 2008 16:26:41 +0000 (16:26 +0000)]
Get the device state of the state interface instead of the interface when creating a new queue member.
Thanks to Atis Lezdins for bringing this up on the Asterisk-Dev mailing list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98993
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Jason Parker [Thu, 17 Jan 2008 16:21:38 +0000 (16:21 +0000)]
Merged revisions 98991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #11784)
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r98991 | qwell | 2008-01-17 10:19:46 -0600 (Thu, 17 Jan 2008) | 4 lines
Add a clarification about the immediate= option of zapata.conf
Issue 11784, patch by klaus3000.
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Kevin P. Fleming [Thu, 17 Jan 2008 16:17:52 +0000 (16:17 +0000)]
major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98990
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Kevin P. Fleming [Thu, 17 Jan 2008 15:04:54 +0000 (15:04 +0000)]
resolve (valid) compiler warning about variable that could be used before being initialized
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98989
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Terry Wilson [Thu, 17 Jan 2008 03:09:32 +0000 (03:09 +0000)]
Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988
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Tilghman Lesher [Thu, 17 Jan 2008 00:13:32 +0000 (00:13 +0000)]
Change the way the new filter feature works, by allowing it to be a column NOT
logged into the database. This will allow more granularity of a decision
evaluated in the dialplan, then takes effect when posting the CDR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98987
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Russell Bryant [Thu, 17 Jan 2008 00:05:13 +0000 (00:05 +0000)]
Add support for an easy way to automatically execute some Asterisk CLI commands
immediately at startup. Any commands in the startup_commands file in the Asterisk
config diretory will get executed.
(closes issue #11781)
Reported by: jamesgolovich
Patches:
asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
-- With some changes by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98986
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Jason Parker [Wed, 16 Jan 2008 23:08:11 +0000 (23:08 +0000)]
Change AST_EXT_TOOL_CHECK to attempt to build against <package>_LIB, per recommendations from Russell.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98985
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Tilghman Lesher [Wed, 16 Jan 2008 22:36:58 +0000 (22:36 +0000)]
Info about res_config_curl
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98984
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Russell Bryant [Wed, 16 Jan 2008 22:36:47 +0000 (22:36 +0000)]
Blocked revisions 98982 via svnmerge
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r98982 | russell | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines
Add an unused pointer to the ast_channel struct. This makes the ast_channel structure
retain the same size as it had in previous 1.4 releases. Also, all of the offsets for
members in the structure are still the same (except for the two pointers that got replaced
for the new spy/whisper architecture.)
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Tilghman Lesher [Wed, 16 Jan 2008 22:20:45 +0000 (22:20 +0000)]
New module res_config_curl (closes issue #11747)
Reported by: Corydon76
Patches:
res_config_curl.c uploaded by Corydon76 (license 14)
20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
Tested by: jmls
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98981
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Russell Bryant [Wed, 16 Jan 2008 21:53:10 +0000 (21:53 +0000)]
Merge the changes from issue #10665 from the team/group/sip_session_timers branch.
This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978
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Joshua Colp [Wed, 16 Jan 2008 20:36:21 +0000 (20:36 +0000)]
Blocked revisions 98973 via svnmerge
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r98973 | file | 2008-01-16 16:34:30 -0400 (Wed, 16 Jan 2008) | 2 lines
Bump up cleancount due to previous commit that changed the channel structure.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98975
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Joshua Colp [Wed, 16 Jan 2008 20:35:49 +0000 (20:35 +0000)]
Blocked revisions 98972 via svnmerge
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r98972 | file | 2008-01-16 16:33:47 -0400 (Wed, 16 Jan 2008) | 2 lines
Replace current spy architecture with backport of audiohooks. This should take care of current known spy issues.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98974
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Jason Parker [Wed, 16 Jan 2008 19:41:25 +0000 (19:41 +0000)]
Partially revert r93898, because it broke the way netsnmp was being detected.
rizzo, do you want to discuss so we can rethink this, or do you have another way?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98971
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Jason Parker [Wed, 16 Jan 2008 18:34:19 +0000 (18:34 +0000)]
Add note about new update.log to CHANGES, by request of jmls and further prodding by jsmith.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98969
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Jason Parker [Wed, 16 Jan 2008 18:06:14 +0000 (18:06 +0000)]
Add logging for 'make update' command (also fixes updates in some places). Issue #11766, initial patch by jmls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98968
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Joshua Colp [Wed, 16 Jan 2008 17:51:52 +0000 (17:51 +0000)]
Merged revisions 98966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 lines
Add missing NULLs at end of two ast_load_realtimes.
(closes issue #11769)
Reported by: tequ
Patches:
chaniax.patch uploaded by dimas (license 88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98967
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Mark Michelson [Wed, 16 Jan 2008 17:21:49 +0000 (17:21 +0000)]
Merged revisions 98964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines
Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
(closes issue #11730)
Reported by: UDI-Doug
Patches:
11730.patch uploaded by putnopvut (license 60)
Tested by: UDI-Doug
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Terry Wilson [Wed, 16 Jan 2008 16:06:06 +0000 (16:06 +0000)]
Make users list static
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98962
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Joshua Colp [Wed, 16 Jan 2008 15:09:37 +0000 (15:09 +0000)]
Merged revisions 98960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 lines
Introduce a lock into the dialing API that protects it when destroying the structure.
(closes issue #11687)
Reported by: callguy
Patches:
11687.diff uploaded by file (license 11)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98961
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Joshua Colp [Wed, 16 Jan 2008 15:04:08 +0000 (15:04 +0000)]
Merged revisions 98958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 lines
Add two more SDP names for ulaw and alaw.
(closes issue #11777)
Reported by: tootai
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98959
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Joshua Colp [Wed, 16 Jan 2008 03:09:11 +0000 (03:09 +0000)]
Merged revisions 98955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines
Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
reinvite-patch.txt uploaded by kebl0155 (license 356)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98956
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Joshua Colp [Wed, 16 Jan 2008 02:30:13 +0000 (02:30 +0000)]
Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups.
(closes issue #10983)
Reported by: jtodd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98954
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Steve Murphy [Wed, 16 Jan 2008 01:35:10 +0000 (01:35 +0000)]
Terry found this problem with running the expr2 parser on OSX. Make the #defines come out the same between the parser & lexer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98953
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Joshua Colp [Wed, 16 Jan 2008 01:17:25 +0000 (01:17 +0000)]
Merged revisions 98951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines
Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693)
Reported by: yzg
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Russell Bryant [Tue, 15 Jan 2008 23:53:28 +0000 (23:53 +0000)]
Merged revisions 98946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines
Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
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Tilghman Lesher [Tue, 15 Jan 2008 23:52:11 +0000 (23:52 +0000)]
Add the "filter" keyword
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98947
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Russell Bryant [Tue, 15 Jan 2008 23:35:29 +0000 (23:35 +0000)]
Clean up something I did for ABI compatability in 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98945
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Russell Bryant [Tue, 15 Jan 2008 23:31:53 +0000 (23:31 +0000)]
Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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Joshua Colp [Tue, 15 Jan 2008 20:10:20 +0000 (20:10 +0000)]
Merged revisions 98934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 lines
Based on the boundary found move over the correct amount.
(closes issue #11750)
Reported by: tasker
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98935
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Joshua Colp [Mon, 14 Jan 2008 22:44:20 +0000 (22:44 +0000)]
Merged revisions 98894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 lines
Accept "; boundary=" not just ";boundary=" in the multipart mixed content type.
(closes issue #11750)
Reported by: tasker
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98895
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Jason Parker [Mon, 14 Jan 2008 22:19:40 +0000 (22:19 +0000)]
Add backupdeleted option to app_voicemail
(closes issue #10740)
Reported by: ruffle
Patches:
app_voicemail.diff uploaded by ruffle (license 201)
10740-voicemail.diff uploaded by qwell (license 4)
20080113_bug10740.diff.txt uploaded by mvanbaak (license 7)
Tested by: blitzrage, mvanbaak, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98889
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Mark Michelson [Mon, 14 Jan 2008 22:11:50 +0000 (22:11 +0000)]
Big improvement for app_directory. This patch breaks the do_directory function up
so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:
"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.
1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."
This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.
Huge thanks to dimas for this wonderful submission.
(closes issue #11744)
Reported by: dimas
Patches:
dir3.patch uploaded by dimas (license 88)
Tested by: putnopvut, dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98888
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Mark Michelson [Mon, 14 Jan 2008 21:01:56 +0000 (21:01 +0000)]
Blocked revisions 98849 via svnmerge
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r98849 | mmichelson | 2008-01-14 14:59:26 -0600 (Mon, 14 Jan 2008) | 4 lines
Adding in appropriate unlocks for the locks I added. Thanks to joetester on IRC
for pointing this out.
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Joshua Colp [Mon, 14 Jan 2008 20:01:19 +0000 (20:01 +0000)]
Make sure the user's manager secret exists, even if it is blank.
(closes issue #11749)
Reported by: srt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98830
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Terry Wilson [Mon, 14 Jan 2008 18:42:16 +0000 (18:42 +0000)]
Add description of TOUPPER and TOLOWER dialplan functions to CHANGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98811
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Jason Parker [Mon, 14 Jan 2008 17:40:36 +0000 (17:40 +0000)]
Add proper call forwarding (all and busy) support for chan_skinny.
Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776
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Russell Bryant [Mon, 14 Jan 2008 17:39:31 +0000 (17:39 +0000)]
Merged revisions 98774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines
Revert a change that introduces an unacceptable performance hit and is causing
memory leaks ... (from rev 97973)
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Jason Parker [Mon, 14 Jan 2008 17:18:35 +0000 (17:18 +0000)]
Fix for potential crash with vmexten
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98773
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Mark Michelson [Mon, 14 Jan 2008 16:36:41 +0000 (16:36 +0000)]
Merged revisions 98737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan 2008) | 3 lines
Fixing another compilation error. I'm a bit off today :(
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Mark Michelson [Mon, 14 Jan 2008 16:34:00 +0000 (16:34 +0000)]
Blocked revisions 98734 via svnmerge
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r98734 | mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3 lines
Oops. Last commit had compilation error.
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Mark Michelson [Mon, 14 Jan 2008 16:33:06 +0000 (16:33 +0000)]
Merged revisions 98733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines
Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.
Thanks to John Bigelow for pointing this error out to me.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98735
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Joshua Colp [Mon, 14 Jan 2008 15:07:30 +0000 (15:07 +0000)]
Print out a warning when spaces are used in the variable name in Set and MSet. It is extremely hard to debug this issue so this should make it easier.
(closes issue #11759)
Reported by: caio1982
Patches:
setvar_space_warning1.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98714
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Joshua Colp [Mon, 14 Jan 2008 14:33:17 +0000 (14:33 +0000)]
Update documentation.
(closes issue #11763)
Reported by: IgorG
Patches:
docupd.v1.diff uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98695
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Russell Bryant [Mon, 14 Jan 2008 04:53:08 +0000 (04:53 +0000)]
Add another small option for the JACK app and JACK_HOOK function. The 'n'
option tells JACK not to start jackd automatically if it is not already
running. Otherwise, the default is that jackd will get started for you if
it isn't running already.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98676
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Russell Bryant [Sun, 13 Jan 2008 23:43:06 +0000 (23:43 +0000)]
- Break up the Misc. section a bit with a new section for Misc. New Modules
- Change spacing a bit in some places for consistent indentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98656
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Russell Bryant [Sun, 13 Jan 2008 19:19:57 +0000 (19:19 +0000)]
Bring in the code from team/russell/jack/.
Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98628
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Russell Bryant [Sun, 13 Jan 2008 18:08:50 +0000 (18:08 +0000)]
Add configure script check for JACK.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605
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Russell Bryant [Sun, 13 Jan 2008 18:01:56 +0000 (18:01 +0000)]
Remove KDE configure script check that isn't used
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98604
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Russell Bryant [Sun, 13 Jan 2008 00:10:00 +0000 (00:10 +0000)]
Remove a duplicate lock of the audiohook lock when destroying manipulate
audiohooks. This causes an error when we attempt to destroy the lock later
when freeing the audiohook.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98581
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Russell Bryant [Sat, 12 Jan 2008 19:34:38 +0000 (19:34 +0000)]
Add a new CLI command, "core set chanvar", which allows you to set a channel
variable (or function) on an active channel from the CLI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98558
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Tilghman Lesher [Sat, 12 Jan 2008 18:12:56 +0000 (18:12 +0000)]
Conversion to load manager.conf into memory did not convert the password
functions correctly. (Closes issue #11749)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98536
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Pari Nannapaneni [Sat, 12 Jan 2008 05:13:04 +0000 (05:13 +0000)]
merging a comment added in 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98514
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Kevin P. Fleming [Sat, 12 Jan 2008 00:20:55 +0000 (00:20 +0000)]
Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated.
(closes issue #11212)
Reported by: tzafrir
Patches:
zap_dnd.diff uploaded by tzafrir (modified by me) (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98488
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Tilghman Lesher [Sat, 12 Jan 2008 00:17:26 +0000 (00:17 +0000)]
Merged revisions 98467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines
Add a connection timeout attribute, as that was what was intended with the
login timeout, but ODBC divides it up into 2 different timeouts.
(Closes issue #11745)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98487
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Russell Bryant [Fri, 11 Jan 2008 23:57:01 +0000 (23:57 +0000)]
Add some extra checking to help out with a potential error when trying to
run "make asterisk.pdf" when not all of the right packages are installed.
(closes issue #10763)
Reported by: Corydon76
Patches:
20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98454
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