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(issue #7941 reported by eclubb)
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setting of frame.delivery fields, some minor code cleanups
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(issue #8065, adamg)
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static member list updating
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statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach)
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(issue #7370 reported by kryptolus)
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reported and patched)
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ahead of time. (Issue 7631)
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Playback() returns -1 on missing args - so should Background()
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feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@43778 f38db490-d61c-443f-a65b-d21fe96a405b
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properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed.
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calling write(). During this change, checking if this was a remote console was
removed. This caused this message about using "exit" or "quit" to exit an
Asterisk console to come up in times where it did not make sense. This change
restores the check to see if this is a remote console before printing the
message. (fixes BE-4)
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don't introduce a length limit on the definition by using a 256 byte temporary
storage buffer. Instead, make the temporary buffer just as big as it needs
to be to hold the entire mailbox definition.
(fixes BE-68)
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1) delete=yes was ignored
2) maxmessages was ignored
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an IAX2 channel before proceeding to process it to avoid crashing.
(issue #8017, reported by admott, fixed by myself)
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cleanup function and a memory allocation issue. (issue #7960 reported by jojo & issue #7999 reported by aster1) Special thanks to csum77 for letting me into a box where this issue was happening.
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(Ugly fix, not proud at all)
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nic_bellamy)
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(according to Directory's own documentation) - Issue 7965
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@43072 f38db490-d61c-443f-a65b-d21fe96a405b
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back to 0 (Issue 6656)
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After that, assume the paging already completed by the time the channel entered
the conference and drop back out. (Issue 7275)
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raw format OR if a translation path already exists to translate between them. (issue #7887 reported by softins & issue #7803 reported by alvaro_palma_aste). Thanks goes to stubert for giving me access to a box and showing me a scenario where this occured.
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- Don't delete SIP dialog prematurely
Strangely enough imported from svn trunk... It's confusing here in Greenland.
(Committing from 36.000 feet above Greenland, on the way to asterisk@von
http://www.pulver.com/asterisk )
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if authentication fails. (Bug #7907)
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good idea (Bug 7811)
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ast_cond_timedwait (issue #7697 reported by bn999)
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getting lost because logging isn't configured yet. The problem was that the code that exists to handle this case was not getting reached, because other tests were causing an early return from ast_log().
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makefile, as per bug 7545
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it will have already been closed elsewhere and could cause massive panic. (issue #7699 reported by bn999)
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As posted from #7458 and others similar to it in Mantis:
p->app_lock was a mutex really designed for use with agents not in callback mode. That being the case, I've tried to code it so that when callback mode is used, the app_lock mutex will not be locked/unlocked at all. Please let me know how you make out - and if you continue to deadlock now, please reproduce the deadlock logging information and post to Mantis.
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log to the very begining of misdn_hangup.
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