solution, it breaks communication.
Rizzo - you need to implement a configuration option for this
code. It's good, but maybe should be off by default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
way of handling DTMF in SIP. Totally undocumented, but implemented
in enough devices so we have to support it.
Code by sergee, small changes by oej.
Closes issue #11049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
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r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line
if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
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r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line
added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it.
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r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line
fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all)
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r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line
aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
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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: https://origsvn.digium.com/svn/asterisk/trunk@89120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@89108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
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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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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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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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- 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.)
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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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r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines
When traversing the list of channel variables here in transmit_invite(), the
asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Christian Richter
Luigi Rizzo
Tilghman Lesher
Joshua Colp
Olle Johansson
Russell Bryant
Jason Parker
Mark Michelson
Kevin P. Fleming
Matthew Fredrickson