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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) | 2 lines
Typo
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines
Do not add a sip: to the beginning of the To URI unless needed.
(closes issue #10756)
Reported by: goestelecom
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines
Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
(closes issue #10164)
Reported by: nic_bellamy
Patches:
sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines
Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support.
(closes issue #10946)
Reported by: flefoll
(closes issue #10915)
Reported by: ramonpeek
(closes issue #9567)
Reported by: atca_pres
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines
If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan.
(closes issue #11185)
Reported by: spditner
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) | 7 lines
The member refcount must be incremented, to avoid using it after deallocation.
A huge thanks go to lvl- for patiently providing the necessary valgrind output
that was necessary to finding this problem of memory corruption.
Reported by: lvl-
Patch by: tilghman
Closes issue #11174
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INBOX...
ever. It may not exist.
(closes issue #11151, reported by selsky, patched by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines
This patch makes it possible for SIP phones to dial extensions defined with '#' characters
in extensions.conf AND maintain their escaped characters when forming URI's
(closes issue #10681, reported by cahen, patched by me, code review by file)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line
In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
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If we really want them we can remove the --no-print-directory
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Reported by: eliel
Patch by: eliel
Closes issue #11177
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) | 5 lines
Suppress AEL warnings on load.
Reported by: eliel
Patch by: eliel
Closes issue #11178
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Reported by: thetatag
Patch by: thetatag/stevens/tilghman
Closes issue #5331
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while also
buffering the audio stream.
Reported by: devil_slayer
Patch by: devil_slayer
Closes issue #8921
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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initialization.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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AGI debugging on busy systems much easier.
(closes issue #10730)
Reported by: junky
Patches:
agi_debug_chan.diff uploaded by junky (license 177)
20070923_10730.diff uploaded by mvanbaak (license 7)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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when it should have been collecting more digits.
(closes issue #11175)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10942, reported and patched by julianjm, documentation changes by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #9374, reported and initially patched by wuwu, intermediate patch by eliel, and final patch by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 line
closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix.
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batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed.
(closes issue #11172)
Reported by: ibc
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call
the device that made the transfer instead. This makes for much smoother calling back
when queues are involved.
(closes issue #11155, reported by IPetrov)
Tremendous thanks to Russell for pulling me out of my block I was having on this one
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 Nov 2007) | 3 lines
Fix init_classes() so that classes that actually do have files loaded aren't
treated as empty, and immediately destroyed ...
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of the current memory allocations when you start Asterisk, when the command's
handler gets called for initialization.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 Nov 2007) | 2 lines
We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines
Correctly set the total number of channels from a zaptel transcoder board.
SPD-49, patch by Matthew Nicholson.
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines
Bug fixes to tdd support in zaptel.
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(Small changes for trunk)
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Closes issue #9972
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r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines
If someone were to delete the files used by an existing MOH class, and then
issue a reload, further use of that class could result in a crash due to
dividing by zero. This set of changes fixes up some places to prevent this
from happening.
(closes issue #10948)
Reported by: jcomellas
Patches:
res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282)
Additional changes added by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines
Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
(closes issue #11006)
Reported by: pj
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the constructor for the list of modules was run
after the constructors for the embedded modules
(which appended entries to the list).
As a result, the list appeared empty when it was
time to use it.
On linux the order of execution of constructor
was evidently different (it may depend on the
ordering of modules in the ELF file).
This is only a workaround - there may be other
situations where the execution of constructors
causes problems, so if we manage to find a more
general solution this workaround can go away.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines
Fix improbable but possible memory leaks in chan_zap.
(closes issue #11166)
Reported by: eliel
Patches:
chan_zap.c.patch uploaded by eliel (license 64)
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way.
(closes issue #11167)
Reported by: eliel
Patches:
chan_agent.c.patch uploaded by eliel (license 64)
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(closes issue #11168)
Reported by: IgorG
Patches:
unistimjb-88863-1.patch uploaded by IgorG (license 20)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
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r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines
Remove some checks to see if locks are initialized from the non-DEBUG_THREADS
versions of the lock routines. These are incorrect for a number of reasons:
- It breaks the build on mac.
- If there is a problem with locks not getting initialized, then the proper
fix is to find that place and fix the code so that it does get initialized.
- If additional debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS wrappers.
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