https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines
Rework of the commit I made yesterday to use the already built-in
ast_uri_decode function as opposed to my home-rolled one. Also added
comments.
Thanks to oej for pointing me in the right direction
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89120 f38db490-d61c-443f-a65b-d21fe96a405b
The type of warnings emitted depends on the optimization level,
at the lower levels the compiler doesn't always understand what the
programmer has in mind. In this case I could not understand it either.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89108 f38db490-d61c-443f-a65b-d21fe96a405b
- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89089 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89073 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89069 f38db490-d61c-443f-a65b-d21fe96a405b
(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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89062 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89055 f38db490-d61c-443f-a65b-d21fe96a405b
of the current memory allocations when you start Asterisk, when the command's
handler gets called for initialization.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89052 f38db490-d61c-443f-a65b-d21fe96a405b
qwell