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r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines
Specify codecs in callfiles and manager, to allow video calls to be set up
from callfiles and AMI.
(closes issue #9531)
Reported by: Geisj
Patches:
20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14)
20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134980 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines
(closes issue #11849)
Reported by: greyvoip
Tested by: murf
OK, a few days of debugging, a bunch of instrumentation
in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid
notebook pages of notes later, I have made the small
tweek necc. to get the start time right on the second
CDR when:
A Calls B
B answ.
A hits Xfer button on sip phone,
A dials C and hits the OK button,
A hangs up
C answers ringing phone
B and C converse
B and/or C hangs up
But does not harm the scenario where:
A Calls B
B answ.
B hits xfer button on sip phone,
B dials C and hits the OK button,
B hangs up
C answers ringing phone
A and C converse
A and/or C hangs up
The difference in start times on the second CDR is because
of a Masquerade on the B channel when the xfer number is
sent. It ends up replacing the CDR on the B channel with
a duplicate, which ends up getting tossed out. We keep
a pointer to the first CDR, and update *that* after the
bridge closes. But, only if the CDR has changed.
I hope this change is specific enough not to muck
up any current CDR-based apps. In my defence, I
assert that the previous information was wrong,
and this change fixes it, and possibly other
similar scenarios.
I wonder if I should be doing the same thing
for the channel, as I did for the peer, but
I can't think of a scenario this might affect.
I leave it, then, as an exersize for the users,
to find the scenario where the chan's CDR
changes and loses the proper start time.
........
and as to 1.4 to trunk; have I expressed my
feelings about code shifting from one file
to another? Good.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134922 f38db490-d61c-443f-a65b-d21fe96a405b
1) If a function returns SQLITE_LOCKED, no recovery is possible.
2) An error message can be allocated, even when no error is signalled.
(closes issue #13109)
Reported by: gknispel_proformatique
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134919 f38db490-d61c-443f-a65b-d21fe96a405b
This ensures that we don't just keep a cache of tiny frames, continually doing
an alloc/free for each data frame, thus negating the point of having a cache.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134867 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) | 7 lines
In case we have some processing threads that free more frames than they allocate,
do not let the frame cache grow forever.
(closes issue #13160)
Reported by: tavius
Tested by: tavius, russell
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134815 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines
Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).
(closes issue #13186)
Reported by: miquel_cabrespina
Patches:
13186.diff uploaded by putnopvut (license 60)
Tested by: miquel_cabrespina
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134759 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) | 4 lines
Qwell pointed out, via IRC, that the previous fix only worked when explicitly
set. When nothing is set, and the option is implied, it breaks, because
configure sets the prefix to 'NONE'. Fixing.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134650 f38db490-d61c-443f-a65b-d21fe96a405b
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
respectively. Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
Reported by: eliel
Patches:
cliprompt.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134353 f38db490-d61c-443f-a65b-d21fe96a405b
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.
On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.
Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.
closes issue #11928)
Reported by: adriavidal
Patches:
1.6.0-configurev2.patch uploaded by putnopvut (license 60)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134125 f38db490-d61c-443f-a65b-d21fe96a405b
called from elsewhere in Asterisk to find the current state of a device. In
that case, we want to use the cached value if it exists. The other way is when
processing a device state change. In that case, we do not want to check the
cache because returning the last known state is counter productive.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133945 f38db490-d61c-443f-a65b-d21fe96a405b
we do NOT need to uri_decode in manager.
(if I sent core%20show%20channels from a telnet
session, it should be interpreted literally, however,
if I send that from an http session, it should be
decoded, which is the behaivor now)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133770 f38db490-d61c-443f-a65b-d21fe96a405b
