monitor a jabber connection over manager.
patches from 7673 and 7666 with minor changes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39248 f38db490-d61c-443f-a65b-d21fe96a405b
post to the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html
This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on. I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39109 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r39081 | russell | 2006-08-06 21:28:29 -0400 (Sun, 06 Aug 2006) | 7 lines
Fix a crash reported to me by hads on IRC. This crash would occur with the use
of the "distinctiveringaftercid" option. Also, on this user's system, the crash
would only occur when built without optimizations. This is because the bug is
that the code would write past the end of an array that was allocated on the
stack, and the structure of the stack is different with or without optimizations
enabled.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39082 f38db490-d61c-443f-a65b-d21fe96a405b
- use appropriate types in some assignments
- use ast_strlen_zero()
- don't manually free cid fields since ast_set_callerid() will handle it
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39058 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38982 | russell | 2006-08-05 05:01:37 -0400 (Sat, 05 Aug 2006) | 6 lines
Always generate a Newstate event in ast_setstate() instead of making it a
Newchannel event if the state was AST_STATE_DOWN. The Newchannel event will
always be generated in ast_request(), so this just causes a duplicated
Newchannel event in some cases.
(issue #7506, repoted by capouch, fixed by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38994 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
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r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38905 f38db490-d61c-443f-a65b-d21fe96a405b
* added blocking flag to stack object. A port can be blocked/unblocked from the
cli
* added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later
we can add a manager event for that)
* added block_on_alarm option, to block the port whenever a ALARM occurs
* added need_busy flag to indicate if we've sended a CONTROL_BUSY already
* changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in
recent asterisk ast_log messages..
* fixed a few ETSI state violations, especially when finishing calls in
different seldom states
* changed debug levels a lot to make the log more readable in low debuglevels
* some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still)
* removed the PRECONNECTED state stuff
* added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this
creates a CHANISUNAVAIL instead of a NOANSWER
* removed the addr pointer from "misdn show stacks" that's not needed anymore
and makes the output more unreadable
* added cause saving on RELEASE/RELEASE_COMPLETE
* set cause to 16 on prepare_bc
* removed stack getting from ph_control functions, we don't really need it
there
* added beroec api
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38801 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38686 | kpfleming | 2006-08-01 18:07:06 -0500 (Tue, 01 Aug 2006) | 2 lines
ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38687 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38654 | file | 2006-08-01 15:20:05 -0400 (Tue, 01 Aug 2006) | 2 lines
Close the stream when file based MOH stop. This won't get rid of their position in the file but it will cause the translation path to be setup again. (issue #7634 reported by asimpson)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38655 f38db490-d61c-443f-a65b-d21fe96a405b
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r38611 | kpfleming | 2006-07-31 16:14:11 -0500 (Mon, 31 Jul 2006) | 4 lines
don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough)
don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others)
use ast_softhangup_nolock() since we already hold the channel's lock
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38612 f38db490-d61c-443f-a65b-d21fe96a405b
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r38585 | file | 2006-07-31 13:09:10 -0400 (Mon, 31 Jul 2006) | 2 lines
Add missing code to bring transferee channel out of MOH/autoservice under certain circumstance (issue #7611 reported by guillecabeza with minor mods by myself)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38586 f38db490-d61c-443f-a65b-d21fe96a405b
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r38546 | russell | 2006-07-31 00:04:02 -0400 (Mon, 31 Jul 2006) | 2 lines
Make the frame counting done with TRACE_FRAMES defined thread-safe
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r38547 | russell | 2006-07-31 00:06:16 -0400 (Mon, 31 Jul 2006) | 2 lines
one more small tweak for thread-safety of TRACE_FRAMES
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38548 f38db490-d61c-443f-a65b-d21fe96a405b
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r38501 | file | 2006-07-29 19:18:00 -0400 (Sat, 29 Jul 2006) | 2 lines
How many attempts does it take to make a SIP URI parser that works well? I'm up to 5 personally. On to the good stuff - parse the domain first, user second, and get rid of port & options/params last. (issue #7616 reported by andrew)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38502 f38db490-d61c-443f-a65b-d21fe96a405b
version check here anwyay, define attribute_pure to be empty if it's an earlier
version.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38464 f38db490-d61c-443f-a65b-d21fe96a405b
int to string or string to int operations.
"pure" essentially says that this function has no side effects aside from its
result, and the result depends on nothing else other than its arguments and
global variables. "const" is a more strict form of "pure", where the function
also doesn't access any global variables.
From the gcc manual: "Such a function can be subject to common subexpression
elimination and loop optimization just as an arithmetic operator would be."
This also tells the compiler that it is safe to call the function fewer times
than the code says to, given the same arguments, since the result will always
be the same.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38452 f38db490-d61c-443f-a65b-d21fe96a405b