https://origsvn.digium.com/svn/asterisk/branches/1.4
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r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines
(closes issue #13236)
Reported by: korihor
Wow, this one was a challenge!
I regrouped and ran a new strategy for
setting the ~~MACRO~~ value; I set it once
per extension, up near the top. It is only
set if there is a switch in the extension.
So, I had to put in a chunk of code to detect
a switch in the pval tree.
I moved the code to insert the set of ~~exten~~
up to the beginning of the gen_prios routine,
instead of down in the switch code.
I learned that I have to push the detection
of the switches down into the code, so everywhere
I create a new exten in gen_prios, I make sure
to pass onto it the values of the mother_exten
first, and the exten next.
I had to add a couple fields to the exten
struct to accomplish this, in the ael_structs.h
file. The checked field makes it so we don't
repeat the switch search if it's been done.
I also updated the regressions.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
trunk.
For an explanation of what "imap_consistency" is,
please see svn revision 134223 to the 1.4 branch.
Coincidentally, this also fixes a recent bug report
regarding the inability to save messages to the new
folder when using IMAP storage since they will would
be flagged as "seen" and not be recognized as new
messages.
(closes issue #13234)
Reported by: jaroth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@136633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@136631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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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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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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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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: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Changing debug messages from VERBOSE to DEBUG channel
- Adding a few todo's
- Adding a few more "XMPP"'s to compliment Jabber...
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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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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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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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