would reset to 500 ms every time a non-voice frame
was received. The total time we poll should be 500 ms, so
now we save the amount of time left after the poll returned
and use that as our argument for the next call to poll
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136633 f38db490-d61c-443f-a65b-d21fe96a405b
Instead, poll the channel until receiving a voice frame. The
cap on this poll is 500 ms.
The optional delay is still allowable in the Answer() application,
but the delay has been moved back to its original position, after
the call to the channel's answer callback. The poll for the voice
frame will not happen if a delay is specified when calling Answer().
(closes issue #12708)
Reported by: kactus
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136631 f38db490-d61c-443f-a65b-d21fe96a405b
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r136560 | kpfleming | 2008-08-07 13:25:31 -0500 (Thu, 07 Aug 2008) | 3 lines
change the required dependency for codec_dahdi to only be satisfied by DAHDI and not Zaptel, as the new transcoder interface is only in DAHDI
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136562 f38db490-d61c-443f-a65b-d21fe96a405b
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r136458 | kpfleming | 2008-08-07 11:11:17 -0500 (Thu, 07 Aug 2008) | 3 lines
work around a bug in gcc-4.2.3 that incorrectly ignores the casting away of 'const' for pointers when the developer knows it is safe to do so
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136481 f38db490-d61c-443f-a65b-d21fe96a405b
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r136404 | kpfleming | 2008-08-07 10:07:12 -0500 (Thu, 07 Aug 2008) | 2 lines
remove config.cache during distclean, in case the user is using autoconf caching
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136405 f38db490-d61c-443f-a65b-d21fe96a405b
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r136304 | tilghman | 2008-08-06 20:17:14 -0500 (Wed, 06 Aug 2008) | 3 lines
For backwards compatibility with previous 1.4 versions which used "zapchan"
in users.conf, ensure that we still support it.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136305 f38db490-d61c-443f-a65b-d21fe96a405b
this in pieces so the diffs are a little bit smaller and more reviewable.
pbx/ and formats/ first.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136298 f38db490-d61c-443f-a65b-d21fe96a405b
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r136238 | mmichelson | 2008-08-06 15:42:15 -0500 (Wed, 06 Aug 2008) | 4 lines
We only need to unregister the QueueStatus manager
command once on an unload
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136239 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines
Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136063 f38db490-d61c-443f-a65b-d21fe96a405b
- Changing debug messages from VERBOSE to DEBUG channel
- Adding a few todo's
- Adding a few more "XMPP"'s to compliment Jabber...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136005 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines
Merging the issue11259 branch.
The purpose of this branch was to take into account
"burps" which could cause jitterbuffers to misbehave.
One such example is if the L option to Dial() were used
to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through
the jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a
brief period.
Now ast_generic_bridge will empty and reset the jitterbuffer
each time it is called. This causes injected audio to be handled
properly.
ast_generic_bridge also will empty and reset the jitterbuffer
if it receives an AST_CONTROL_SRCUPDATE frame since the change
in audio source could negatively affect the jitterbuffer.
All of this was made possible by adding a new public API call
to the abstract_jb called ast_jb_empty_and_reset.
(closes issue #11259)
Reported by: plack
Tested by: putnopvut
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r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines
Revert inadvertent changes to app_skel that occurred when
I was testing for a memory leak
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r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines
Remove properties that should not be here
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135851 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135821 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines
In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID
was perverted. This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank. In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135748 f38db490-d61c-443f-a65b-d21fe96a405b
never indicate a match, so nothing would have been returned anyway, but it was
still a poor example of proper usage.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135439 f38db490-d61c-443f-a65b-d21fe96a405b
