ast_rtp_new_with_bindaddr():
1. add comments to the logic of the main loop;
2. use a common exit point on failure so the cleanup is done only in one place;
3. handle failures in rtp_socket() in the main loop of the function;
No functional changes except for #3 above, so it is not yet
worthwhile merging this and other changes to 1.4
Once the cleanup work on this file will be complete (which among
other things should include some extensions to the stun support)
it might be a good thing to push all the changes to 1.4
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r74767 | russell | 2007-07-11 17:57:07 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines
The function make_trunk() can fail and return -1 instead of a valid new call
number. Fix the uses of this function to handle this instead of treating it
as the new call number. This would cause a deadlock and memory corruption.
(possible cause of issue #9614 and others, patch by me)
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(which is very little, at the moment).
Eventually, when the functionality is extended, the changes can be merged
back to 1.4. At the moment this is pointless.
Note, this change is whitespace only.
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(closes issue #10158)
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r74428 | qwell | 2007-07-10 14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines
Merged revisions 74427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 lines
Fix an issue where it was possible to have a service level of over 100%
Between the time recalc_holdtime and update_queue was called, it was possible that the call could have been hungup.
Move both additions to the same place, so this won't happen.
Issue 10158, initial patch by makoto, modified by me.
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(closes issue #10133)
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r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines
Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines
Use res_ndestroy on systems that have it. Otherwise, use res_nclose.
This prevents a memleak on NetBSD - and possibly others.
Issue 10133, patch by me, reported and tested by scw
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r74262 | file | 2007-07-10 11:07:13 -0300 (Tue, 10 Jul 2007) | 2 lines
Only spit out an inringing warning message when it is applicable. Since call limits are already toast in realtime let's not scare the user if they are using it. (issue #10166 reported by bcnit)
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r74211 | russell | 2007-07-09 16:31:30 -0500 (Mon, 09 Jul 2007) | 5 lines
Update the configure script to check for a required function that is not present
in the 1.2 version of libpri. This will prevent the configure script from thinking
that it has compatible libpri support for Asterisk 1.4, when it actually does not
because the installed version is from 1.2.
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Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
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r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09 Jul 2007) | 9 lines
(closes issue #10123)
Reported by: blitzrage
Patches submitted by: juggie, qwell, me
Tested by: blitzrage
When trying to find a music on hold class to use, try all of the options,
instead of only the first one that is set. Also, change the MusicOnHold
applications to not hang up on the channel when a class can not be found.
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Closes issue #9186
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r74159 | qwell | 2007-07-09 15:19:28 -0500 (Mon, 09 Jul 2007) | 16 lines
Merged revisions 74158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines
Several chan_zap options were not working on reload because they were arbitrarily
disallowed when reloading some/most PRI options (such as signalling) was disallowed.
Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload.
This corrects that behavior.
Issue 9186, patch by tzafrir.
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r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul 2007) | 6 lines
The n option for Queue should make the queue exit immediately after failure to reach any members and should not
be dependent on the timeout value passed to Queue
(closes issue #10127, reported by bcnit, repaired by me)
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If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
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