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r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines
Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote.
(closes issue #11027)
Reported by: ramonpeek
Patches:
11027-1.diff uploaded by ramonpeek (license 266)
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r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | 10 lines
The channel needs to stay locked while running timer callbacks, as they access
and modify channel data that may change elsewhere. I went through every timer
callback in the source tree to make sure that none of them did any additional
locking that could introduce deadlocks, and all is well.
(closes issue #10765)
Reported by: Ivan
Patches:
ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229)
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r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct 2007) | 5 lines
If a non-existent file is specified to be played either as a periodic announcement
or as a hold/position announcement, the caller would be kicked out of the queue.
No longer does this happen.
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r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) | 9 lines
Revert a change that I made for issue #10979 which, as has been pointed out to
me in issue #11018, doesn't really make sense. There is no reason to have
the base64 decode function force a '\0' terminated buffer, when the result is
almost always binary, anyway. In fact, this caused some breakage, as some code
in res_crypto passed in a buffer exactly the right size to get its binary
result, which got stomped on by this patch.
(closes issue #11018, reported by dimas)
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r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct 2007) | 6 lines
Changing the strategy field of the call_queue struct to be signed instead of unsigned,
since the code attempts to set the strategy to -1 if you specify a bogus strategy.
While this isn't a huge issue in 1.4, it could be a problem for someone who, say, tries
to use the roundrobin strategy in trunk (despite all the deprecation warnings in 1.4).
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r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines
If Asterisk is in the middle of shutting down, respond to OPTIONS
with 503 Unavailable.
(closes issue #10994)
Reported by: eserra
Patches:
sip-options-503.patch uploaded by eserra (license 45)
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r86028 | mmichelson | 2007-10-16 17:49:10 -0500 (Tue, 16 Oct 2007) | 6 lines
Adding deprecated warning to monitor-join option, since the plan is to
no longer support this in favor of monitor-type = mixmonitor
(related to issue #10885)
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r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 Oct 2007) | 16 lines
Some locking errors exposed the fact that the lock debugging code itself was
not thread safe. How ironic! Anyway, these changes ensure that the code that
is accessing the lock debugging data is thread-safe.
Many thanks to Ivan for finding and fixing the core issue here, and also
thanks to those that tested the patch and provided test results.
(closes issue #10571)
(closes issue #10886)
(closes issue #10875)
(might close some others, as well ...)
Patches: (from issue #10571)
ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229)
- a few small changes by me
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r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct 2007) | 5 lines
Trying to remove a non-dynamic queue member via dynamic means can lead to some
interesting (read nasty) situations. This patch clears up the issue by making
only dynamic queue members removable via dynamic methods.
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Don't improperly memset() over an ast_str. This was leftover from before it
got changed to use ast_str.
(closes issue #11003, reported by pj)
(closes issue #10770, reported by yehavi)
(patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.
Now we have Jingle audio, at least between two Asterisk Jingle
clients.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
TransNexus OSP Development
Jason Parker
Russell Bryant
Mark Michelson
Tilghman Lesher
Steve Murphy
Matthew Fredrickson
Philippe Sultan