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- Remove "musiconhold" and "musicclass" settings (also deprecated earlier)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) | 2 lines
make the option actually match the documentation
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) | 2 lines
allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) | 3 lines
Fix a typo in an error message that indicated that the MGCP channel type could
not be registered, instead of the correct type, OSS.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48944 | russell | 2006-12-24 02:25:38 -0500 (Sun, 24 Dec 2006) | 11 lines
Merged revisions 48943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines
Check for the proper return value on an error in a call to mmap().
This was reported by Andy Wang on the asterisk-dev list. Thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48940 | russell | 2006-12-24 01:49:31 -0500 (Sun, 24 Dec 2006) | 11 lines
Merged revisions 48939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines
Remove a couple of misplaced dots in log messages. This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec 2006) | 2 lines
Note to self: Run make before committing...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec 2006) | 2 lines
Fix for issue 7774 - patch by alamantia
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compile under --enable-dev-mode
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config files
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006) | 2 lines
suppress compiler warnings in this module until it can be improved
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thread count.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines
Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2 lines
Hold call structure lock in places where a qualify or peer action can destroy it.
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r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2 lines
Lock network retransmission queue in all places that it is used.
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r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2 lines
Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted.
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wrong module.
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just failing to compile.
It seems like the proper way to do this would be in the configure script.
However, that wouldn't help existing checkouts unless we forced the configure
script to be executed after any code was changed.
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r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat, 09 Dec 2006) | 9 lines
Merged revisions 48371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line
This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | 8 lines
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
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frames. The function, before this commit, was roughly 1400 lines long. So, I
am working on breaking this up into functions so that the code is easier to
follow and debug. Also, I will be committing these changes in chunks as I do
them to ease tracking down any potentially introduced problems.
Break out roughly 150 lines from socket_process() and introduce a new function,
socket_process_meta() which handles the parsing of an incoming meta frame.
Also, restructure some of this code a bit to reduce the deep nesting that was
in this code.
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- Use sizeof() to pass an array size to a function
- Use a single bit for a variable in the chan_iax2_pvt stuct since that is all
it needs.
- Add some comments about the iaxs, iaxl, and lastused arrays.
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Reported by Gunnar at Omnitor.
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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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