Russell Bryant [Sat, 5 Jan 2008 22:04:08 +0000 (22:04 +0000)]
Print out the name of a function being registered in color, just like the name
of applications when they get registered.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96716
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Russell Bryant [Sat, 5 Jan 2008 21:35:54 +0000 (21:35 +0000)]
Add a note about changing modules.conf since another console channel driver is
now present that can not be used at the same time as chan_alsa or chan_oss.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96715
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Russell Bryant [Sat, 5 Jan 2008 21:25:45 +0000 (21:25 +0000)]
Add the URL to the home page for portaudio. Also add the location of the
svn repository to check out portaudio v19.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96692
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Russell Bryant [Sat, 5 Jan 2008 02:12:10 +0000 (02:12 +0000)]
Merged revisions 96644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | 2 lines
Don't pass an empty string as the device name.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96645
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Kevin P. Fleming [Sat, 5 Jan 2008 01:05:50 +0000 (01:05 +0000)]
improve chan_usbradio to use indications just like chan_alsa/chan_oss do now
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96621
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Tilghman Lesher [Fri, 4 Jan 2008 23:12:43 +0000 (23:12 +0000)]
Merged revisions 96575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines
Fix the problem of notification of a device state change to a device with a '-'
in the name. Could probably do with a better fix in trunk, but this bug has
been open way too long without a better solution.
Reported by: stevedavies
Patch by: tilghman
(Closes issue #9668)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96576
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Jason Parker [Fri, 4 Jan 2008 22:57:56 +0000 (22:57 +0000)]
Merged revisions 96573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #11237)
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r96573 | qwell | 2008-01-04 16:55:56 -0600 (Fri, 04 Jan 2008) | 4 lines
Properly continue in the dialplan if using PARKINGEXTEN and the slot is full.
Issue 11237, patch by me.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96574
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Tilghman Lesher [Fri, 4 Jan 2008 19:35:00 +0000 (19:35 +0000)]
Merged revisions 96525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) | 4 lines
If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie.
Reported and patched by: one47
(Closes issue #11535)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96547
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Kevin P. Fleming [Fri, 4 Jan 2008 17:21:25 +0000 (17:21 +0000)]
[commit message]
(closes issue #10393)
Reported by: tzafrir
Patches:
chan_alarm_asterisk.diff uploaded by tzafrir (license 46) (modified by me and added configure script support)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96500
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Philippe Sultan [Fri, 4 Jan 2008 17:19:25 +0000 (17:19 +0000)]
Use SASL DIGEST-MD5 authentication over unsecured network connections only.
This authentication mechanism is implemented under the iksemel API, which
makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing
dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
http://ietfreport.isoc.org/ids-wg-sasl.html.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96499
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Russell Bryant [Fri, 4 Jan 2008 16:21:08 +0000 (16:21 +0000)]
Merged revisions 96449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
Make use of the temporary channel pointer while the pvt is unlocked.
(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96450
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Kevin P. Fleming [Thu, 3 Jan 2008 23:14:15 +0000 (23:14 +0000)]
we have to *always* use a completely silent 'make' invocation for generating the module embedding rules
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96398
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Kevin P. Fleming [Thu, 3 Jan 2008 23:01:53 +0000 (23:01 +0000)]
there was no reason to add this define for non-Solaris platforms
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96397
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Russell Bryant [Thu, 3 Jan 2008 22:46:37 +0000 (22:46 +0000)]
Merged revisions 96394 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | 3 lines
Don't crash if the iax2 pvt structure has been destroyed before we get to this point
(closes issue #11672, reported by snuffy, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96395
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Tilghman Lesher [Thu, 3 Jan 2008 21:58:52 +0000 (21:58 +0000)]
Document recent API addition
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96368
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Tilghman Lesher [Thu, 3 Jan 2008 21:43:58 +0000 (21:43 +0000)]
Merged revisions 96318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03 Jan 2008) | 4 lines
Missed initialization caused crash.
Reported and fixed by: tiziano
(Closes issue #11671)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96332
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Tilghman Lesher [Thu, 3 Jan 2008 21:27:57 +0000 (21:27 +0000)]
Allow the uniqueid to be used for searching for a channel in the list.
Reported and initially patched by: michael-fig
(Closes issue #11340)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96301
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Kevin P. Fleming [Thu, 3 Jan 2008 20:04:30 +0000 (20:04 +0000)]
add some simple infrastructure for modules to be used for testing parts of Asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96272
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Kevin P. Fleming [Thu, 3 Jan 2008 19:56:29 +0000 (19:56 +0000)]
eliminiate sound_thread() and other stuff from chan_oss since Asterisk indications can handle it
remove gentone and all the headers containing tones that are no longer needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96270
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Kevin P. Fleming [Thu, 3 Jan 2008 18:10:13 +0000 (18:10 +0000)]
coding guidelines cleanup
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96245
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Christian Richter [Thu, 3 Jan 2008 14:47:30 +0000 (14:47 +0000)]
Merged revisions 96198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line
when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96221
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Tilghman Lesher [Thu, 3 Jan 2008 06:16:48 +0000 (06:16 +0000)]
Add coordination between AMI and AGI applications, with an asyncagi method
Feature proposed and patched by: moy
(Closes issue #11282)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96174
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Tilghman Lesher [Thu, 3 Jan 2008 01:59:27 +0000 (01:59 +0000)]
Compatibility fix for OpenBSD
Report and fix by: mvanbaak
(Closes issue #11669)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96147
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Mark Michelson [Wed, 2 Jan 2008 23:48:43 +0000 (23:48 +0000)]
Merged revisions 96102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines
We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that
multiple members can have the same name, since the variable was not reset on each iteration of the loop.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96103
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Russell Bryant [Wed, 2 Jan 2008 23:22:25 +0000 (23:22 +0000)]
Add support for generating a ringing sound on an incoming call. This is a bit
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96079
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Russell Bryant [Wed, 2 Jan 2008 22:52:13 +0000 (22:52 +0000)]
Note that this module doesn't actually play a ringing sound for an incoming call
... oops
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96077
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Russell Bryant [Wed, 2 Jan 2008 22:50:09 +0000 (22:50 +0000)]
Show the correct CLI command to answer the call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96076
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Kevin P. Fleming [Wed, 2 Jan 2008 22:41:23 +0000 (22:41 +0000)]
actually parse and store echocan parameters from zapata.conf... this *should* work <G>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96073
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Joshua Colp [Wed, 2 Jan 2008 22:40:55 +0000 (22:40 +0000)]
Don't use AST_C_DEFINE_CHECK for the two pthread things that may not actually be definitions, they could be enums for example.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96071
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Mark Michelson [Wed, 2 Jan 2008 22:29:15 +0000 (22:29 +0000)]
Add curly braces around a compound if statement so that trunk will build properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96028
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Russell Bryant [Wed, 2 Jan 2008 22:15:58 +0000 (22:15 +0000)]
Blocked revisions 96024 via svnmerge
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r96024 | russell | 2008-01-02 16:14:28 -0600 (Wed, 02 Jan 2008) | 2 lines
Convert locks of the contexts list in pbx_config to the appropriate rdlock or wrlock
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96025
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Russell Bryant [Wed, 2 Jan 2008 22:05:31 +0000 (22:05 +0000)]
Blocked revisions 96022 via svnmerge
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r96022 | russell | 2008-01-02 16:04:47 -0600 (Wed, 02 Jan 2008) | 2 lines
pbx_dundi only needs a rdlock on the contexts list.
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Russell Bryant [Wed, 2 Jan 2008 22:00:50 +0000 (22:00 +0000)]
Blocked revisions 96020 via svnmerge
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r96020 | russell | 2008-01-02 16:00:21 -0600 (Wed, 02 Jan 2008) | 2 lines
app_macro only needs a rdlock on the contexts list.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96021
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Kevin P. Fleming [Wed, 2 Jan 2008 21:51:37 +0000 (21:51 +0000)]
another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl if it is present, but doesn't parse any supplied parameters yet
(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96019
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Russell Bryant [Wed, 2 Jan 2008 21:49:44 +0000 (21:49 +0000)]
Add doxygen documentation to libresample.h while it's still fresh on my mind
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96018
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Mark Michelson [Wed, 2 Jan 2008 21:08:33 +0000 (21:08 +0000)]
Change instances of AST_NONSTANDARD_APP_ARGS(foo, bar, ',') to AST_STANDARD_APP_ARGS(foo, bar)
(closes issue #11668, reported and patched by mvanbaak)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95994
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Joshua Colp [Wed, 2 Jan 2008 20:26:25 +0000 (20:26 +0000)]
Merged revisions 95946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines
Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001)
(closes issue #11637)
Reported by: greyvoip
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95947
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Mark Michelson [Wed, 2 Jan 2008 20:23:23 +0000 (20:23 +0000)]
Since ',' is the standard argument separator in trunk, change app_queue
to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining
member data.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95945
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Mark Michelson [Wed, 2 Jan 2008 20:19:40 +0000 (20:19 +0000)]
Fix a typo in a comment. AST_STANDARD_APP_ARGS uses ',' as the separator,
not '|'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95944
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Kevin P. Fleming [Wed, 2 Jan 2008 19:47:25 +0000 (19:47 +0000)]
clean up hwgain CLI command and improve docs for swgain CLI command
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95939
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Kevin P. Fleming [Wed, 2 Jan 2008 19:34:33 +0000 (19:34 +0000)]
improve AC_C_DEFINE_CHECK to not try to evaluate the macro being checked for, but just check for its existence
finish implementation of check for Zaptel HWGAIN support
add check for Zaptel ECHOCANCEL_PARAMS support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95937
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Kevin P. Fleming [Wed, 2 Jan 2008 18:21:04 +0000 (18:21 +0000)]
and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95894
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Kevin P. Fleming [Wed, 2 Jan 2008 18:17:15 +0000 (18:17 +0000)]
umm... this did not compile on x86-64, and could not possibly have worked on any platform as it was passing string pointers to a function expecting ints
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95893
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Mark Michelson [Wed, 2 Jan 2008 18:05:57 +0000 (18:05 +0000)]
Merged revisions 95890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines
A change to improve the accuracy of queue logging in the case where a member does not
answer during the specified timeout period. Prior to this change, there was a small chance
that the member name recorded in this case would be blank. Also prior to this change, if using
the ringall strategy, if no one answered the call during the specified timeout, the member name
listed in the queue log would randomly be one of the members that was rung.
(closes issue #11498, reported and tested by hloubser, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95891
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Jason Parker [Wed, 2 Jan 2008 17:38:02 +0000 (17:38 +0000)]
Update osplookup documentation to use commas instead of pipes.
Closes issue #11666, patch by Laureano.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95888
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Russell Bryant [Wed, 2 Jan 2008 16:20:26 +0000 (16:20 +0000)]
For some odd reason, the last set of libresample build changes from Kevin did
not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95864
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Kevin P. Fleming [Wed, 2 Jan 2008 14:53:26 +0000 (14:53 +0000)]
fix some long-time breakage that kept chan_misdn from being embedded
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95841
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Kevin P. Fleming [Wed, 2 Jan 2008 14:50:46 +0000 (14:50 +0000)]
use the proper technique for including submodules so that embedding will work
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95840
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Kevin P. Fleming [Wed, 2 Jan 2008 14:37:50 +0000 (14:37 +0000)]
note that chan_console requires portaudio v19
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95839
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Kevin P. Fleming [Wed, 2 Jan 2008 14:20:46 +0000 (14:20 +0000)]
actually check for a function present in libiconv (don't know how this test could have worked before) and don't do the check on Linux/GNU systems because libiconv is not present there and attempting to link with '-liconv' always fails (it's not necessary as the iconv functionality is always available)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95817
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Kevin P. Fleming [Wed, 2 Jan 2008 14:05:30 +0000 (14:05 +0000)]
go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95816
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Philippe Sultan [Wed, 2 Jan 2008 11:34:26 +0000 (11:34 +0000)]
Set stream flags to zero upon initialization.
When the XMPP over TLS/SSL connection resets for some reason, it is
wrongly believed as being secured, which makes the re-connection
process endlessly fail. This was reported by mvanbaak in issue #11644.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95794
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Luigi Rizzo [Wed, 2 Jan 2008 09:16:17 +0000 (09:16 +0000)]
some cleanup of this code while I am trying to debug a problem with
gdb dying while debugging asterisk. The problem seems to be related
with a race in the handling of module_list, which in turn is triggeded
by calling dlopen() on a system which uses initializers to create
locks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95772
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Luigi Rizzo [Wed, 2 Jan 2008 08:53:16 +0000 (08:53 +0000)]
There are three instances of the module definition macros,
which make maintaining this file very error prone.
This commit merges the embedded and !embedded versions,
and fixes the C++ version. Eventually we should move to
a single version of the macro.
Too bad C++ doesn't like the C-style struct initializers
.foo = some_value
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95771
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Russell Bryant [Wed, 2 Jan 2008 04:33:53 +0000 (04:33 +0000)]
Don't make libresample print out debugging output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95746
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Russell Bryant [Wed, 2 Jan 2008 04:31:23 +0000 (04:31 +0000)]
Make the translation table show slin16
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95735
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Russell Bryant [Wed, 2 Jan 2008 03:36:39 +0000 (03:36 +0000)]
fix a spacing issue introduced in revision 95443.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95723
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Russell Bryant [Wed, 2 Jan 2008 01:00:44 +0000 (01:00 +0000)]
Instead of linking libresample into the main Asterisk binary, build it as
res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697
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Luigi Rizzo [Tue, 1 Jan 2008 23:55:19 +0000 (23:55 +0000)]
call directly the cli command to implement hangup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95673
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Luigi Rizzo [Tue, 1 Jan 2008 23:54:40 +0000 (23:54 +0000)]
prevent a panic when destroying a channel with no incoming video.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95672
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Luigi Rizzo [Tue, 1 Jan 2008 23:53:43 +0000 (23:53 +0000)]
remove a leftover sleep(1) used for debugging
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Joshua Colp [Tue, 1 Jan 2008 23:09:32 +0000 (23:09 +0000)]
Fix building of codec_resample on platforms other then Cygwin. On everything else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95648
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Luigi Rizzo [Tue, 1 Jan 2008 22:21:39 +0000 (22:21 +0000)]
make codec_resample build on __CYGWIN__, and make it load on FreeBSD
(and probably other systems as well).
Both need libresample.a to be specified in the linking phase,
and cygwin needs <float.h> as other BSD.
The checks for OS-specific headers should really be moved to some
common header though.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95625
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Luigi Rizzo [Tue, 1 Jan 2008 22:08:32 +0000 (22:08 +0000)]
implement "configure" checks for libiconv, and add the
iconv dependency for func_iconv.
This fixes some build issues on CYGWIN and FreeBSD and probably
other platforms where libiconv is not there by default
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95624
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Mark Michelson [Mon, 31 Dec 2007 23:44:45 +0000 (23:44 +0000)]
Merged revisions 95577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
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Russell Bryant [Mon, 31 Dec 2007 22:41:39 +0000 (22:41 +0000)]
Use float.h to fix the build on FreeBSD. Also, add some other platforms as
they are likely the same.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95550
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Russell Bryant [Mon, 31 Dec 2007 21:33:45 +0000 (21:33 +0000)]
Update chan_console to natively use a 16 kHz sample rate. If it is talking
to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95527
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Russell Bryant [Mon, 31 Dec 2007 21:22:31 +0000 (21:22 +0000)]
Merge changes from team/russell/codec_resample
This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501
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Tilghman Lesher [Mon, 31 Dec 2007 20:33:21 +0000 (20:33 +0000)]
Merged revisions 95470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007) | 3 lines
Allow the default "0" to be returned if the STAT fails
(Closes issue #11659)
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Mark Michelson [Mon, 31 Dec 2007 18:46:12 +0000 (18:46 +0000)]
Fix a compiler warning
(closes issue #11658, reported and patched by eliel)
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Russell Bryant [Mon, 31 Dec 2007 16:13:26 +0000 (16:13 +0000)]
Merge the main set of changes from team/russell/chan_console.
Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95412
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Russell Bryant [Mon, 31 Dec 2007 15:55:22 +0000 (15:55 +0000)]
fix a spelling error in a comment
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Russell Bryant [Mon, 31 Dec 2007 15:53:11 +0000 (15:53 +0000)]
Add CV_STRINGFIELD() macro. This lets you set a config variable to a string field.
(from team/russell/chan_console)
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Russell Bryant [Mon, 31 Dec 2007 15:21:15 +0000 (15:21 +0000)]
Regenerate configure script to include check for portaudio.
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Russell Bryant [Mon, 31 Dec 2007 15:20:30 +0000 (15:20 +0000)]
Add configure script checking for portaudio.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95383
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Luigi Rizzo [Sat, 29 Dec 2007 02:02:03 +0000 (02:02 +0000)]
Move grabbers definitions to a separate file, vgrabbers.c, so it is easier
to add more entries. This required moving struct grab_desc to the common
header, and adding an entry in the Makefile.
On passing, cleanup some comments and file headers (some are still missing).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95313
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Luigi Rizzo [Sat, 29 Dec 2007 01:10:14 +0000 (01:10 +0000)]
virtualize the interface for video grabbers, which should
make it easier to add support for more grabbers (V4L2,
firewire, and so on).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95288
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Luigi Rizzo [Fri, 28 Dec 2007 23:15:11 +0000 (23:15 +0000)]
Add a few entries up to 1408x1152 in the table of known video resolutions.
This makes it very convenient to enlarge images using the right-click
on the video window.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95264
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Luigi Rizzo [Fri, 28 Dec 2007 23:04:04 +0000 (23:04 +0000)]
change the interface of video encapsulation routines, they only
need the buffer and mtu as input.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95263
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Luigi Rizzo [Fri, 28 Dec 2007 22:37:40 +0000 (22:37 +0000)]
various rearrangements and renaming of console_video stuff
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Mark Michelson [Fri, 28 Dec 2007 18:39:29 +0000 (18:39 +0000)]
The diff for this change looks really bad, but all I did here was decrease the indentation of most
of the queue_exec function by reversing the logic of an if statement. This change makes the function
comply better with the coding guidelines. Since this change is purely a cosmetic change to the code, I am
only committing the change to trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95233
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Russell Bryant [Fri, 28 Dec 2007 18:26:26 +0000 (18:26 +0000)]
Merged revisions 95191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines
Remove duplicate increment of the header count in the add_header() function.
(closes issue #11648)
Reported by: makoto
Patch provided by sergee, committed patch by me, inspired by comments from putnopvut
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Mark Michelson [Fri, 28 Dec 2007 16:12:06 +0000 (16:12 +0000)]
Some changes to app_amd.
The channel name is printed in verbose messages
maximumWordLength option added.
Duration of words that do not meet the minimum word duration will be logged
The duration of pre-greeting silence will be logged
Only consider us in the greeting if we actually detected a valid word duration.
(closes issue #11650, reported and patched by davevg)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95167
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Luigi Rizzo [Fri, 28 Dec 2007 08:57:01 +0000 (08:57 +0000)]
fix a small bug in printing out geometries - wrong input.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95139
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Mark Michelson [Fri, 28 Dec 2007 00:17:41 +0000 (00:17 +0000)]
Merged revisions 95095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec 2007) | 8 lines
I found a bug while browsing the queue code and managed to reproduce it in a small setup.
If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to
make app_queue think that all members at that penalty level were unavailable and cause the members at the
next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members
at a given penalty level are unreachable.
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Luigi Rizzo [Thu, 27 Dec 2007 23:32:18 +0000 (23:32 +0000)]
remove more unnecessary casts for NULL.
main/say.c is a big offender in this respect.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95073
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Jason Parker [Thu, 27 Dec 2007 23:28:01 +0000 (23:28 +0000)]
Fix -s socket option, and document it as well.
Closes issue #11645, patch by Laureano.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95070
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Luigi Rizzo [Thu, 27 Dec 2007 23:13:15 +0000 (23:13 +0000)]
NULL does not need to be cast to (char *)
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Luigi Rizzo [Thu, 27 Dec 2007 23:00:55 +0000 (23:00 +0000)]
remove useless casts
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95068
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Russell Bryant [Thu, 27 Dec 2007 21:41:22 +0000 (21:41 +0000)]
Merged revisions 95024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines
Don't report a syntax error when an empty string is passed to ast_get_group.
Just return 0.
(closes issue #11540)
Reported by: tzafrir
Patches:
group_empty.diff uploaded by tzafrir (license 46)
-- slightly changed by me
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Mark Michelson [Thu, 27 Dec 2007 20:11:20 +0000 (20:11 +0000)]
Merged revisions 94977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec 2007) | 3 lines
Fixing a typo in a comment.
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Joshua Colp [Thu, 27 Dec 2007 17:34:00 +0000 (17:34 +0000)]
Merged revisions 94924 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines
Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one.
(closes issue #11585)
Reported by: sobomax
Patches:
chan_h323.c.diff uploaded by sobomax (license 359)
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Joshua Colp [Thu, 27 Dec 2007 17:28:52 +0000 (17:28 +0000)]
Merged revisions 94905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4 lines
Use ast_strlen_zero to see if our_contact is set or not on the dialog. It is possible for it to be a pointer to NULL.
(closes issue #11557)
Reported by: FuriousGeorge
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Luigi Rizzo [Thu, 27 Dec 2007 17:26:16 +0000 (17:26 +0000)]
more localization of gui stuff
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94904
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Mark Michelson [Thu, 27 Dec 2007 17:18:39 +0000 (17:18 +0000)]
Adding documentation for new manager actions and events in app_queue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94903
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Luigi Rizzo [Thu, 27 Dec 2007 16:51:08 +0000 (16:51 +0000)]
clarify the type of video support in chan_oss
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Russell Bryant [Thu, 27 Dec 2007 16:11:41 +0000 (16:11 +0000)]
I went looking for where we downloaded the g722 implementation and came across
these two links. So, I'm adding them so they are available for reference later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94877
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Russell Bryant [Thu, 27 Dec 2007 15:17:31 +0000 (15:17 +0000)]
Blocked revisions 94831 via svnmerge
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r94831 | russell | 2007-12-27 09:16:56 -0600 (Thu, 27 Dec 2007) | 5 lines
Now that the contexts lock is a read/write lock, it should not be locked here
in ast_hint_state_changed(). This makes it get locked recursively which now
causes a deadlock.
(closes issue #11080, thanks to callguy for the access to a deadlocked machine)
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Russell Bryant [Thu, 27 Dec 2007 14:52:07 +0000 (14:52 +0000)]
Merged revisions 94828-94829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines
Change ast_translator_best_choice() to only pay attention to audio formats.
This fixes a problem where Asterisk claims that a translation path can not be
found for channels involving video.
(closes issue #11638)
Reported by: cwhuang
Tested by: cwhuang
Patch suggested by cwhuang, with some additional changes by me.
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r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines
Use the constant that I really meant to use here ...
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Olle Johansson [Thu, 27 Dec 2007 09:13:07 +0000 (09:13 +0000)]
This function checks more than just contexts...
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