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make the format explicit in a debug message;
print the audio instead of aggregated peer capability in a debugging msg.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly.
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writing to the wrong byte. Also, remove some non-thread safe test code.
(closes issue #11317)
Reported by: IgorG
Patches:
unistim-2.patch uploaded by IgorG (license 20)
- additional changes by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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who really need it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11311)
Reported by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Closes issue #11312
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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channels/calls/dialogs/whatever.
Closes issue #11312
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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with the SIPPEER() argument of the same name. The deprecation procedure is not
being used here since this is a trunk-only option.
(closes issue #11307, reported by pj, patched by me)
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deprecate the old ':' delimiter
Reported by: pj
Patch by: tilghman
Closes issue #11305
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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were included almost everywhere.
Remove some of the instances.
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With this, you can control the peer in the dialplan, so you avoid placing outbound
calls when the device has reached busy-level.
Reported by pj.
Closes bug #11180
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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working with
the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't
have the proper context set for the user/peer.
However, we might not want to process an authentication for every OPTIONS, so we could
have a config option for this, "optionsforceok" to always answer 200 OK on the request
and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request,
it doesn't care about the reply. Some devices use OPTIONs to discover capabilities,
since we should answer like an INVITE from the device and we need to support that properly
too, which we don't today.
So much to do :-)
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(this should be handled with the dependencies generated by
configure and menuselect, but will be fixed later)
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
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but make sure that asterisk/compiler.h is included everywhere
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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so it can be reused in the implementation of cli commands when
they have a similar syntax.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 Nov 2007) | 2 lines
Fix an uninitialized memory read found by valgrind
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) | 5 lines
Yet another memory corruption issue.
Reported by: atis
Patch by: tilghman
Fixes issue #10923
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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constification, and marking a function in chan_sip as purposely unused until it is fixed up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines
Don't send re-invites during pending INVITE transactions.
Patch by one47 - thanks!
Closes issue #9305
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines
Improve support for multipart messages. Code by gasparz, changes
by me (mostly formatting). Thanks, gasparz!
Closes issue #10947
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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
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ast_variable_new parameters
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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.
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- Fix a minor spelling error in a comment
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines
If we set a value for qualify, we should actually pay attention to it, instead of overriding the value
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines
Fix two cases of memory corruption caused by background threads.
Reported by: atis
Patch by: tilghman
Fixes issue #10923
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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(closes issue #11218)
Reported by: ys
Patches:
trunk89169.diff uploaded by ys (license 281)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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