- Keep RTP running during T.38 session
We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails
later on in the code, but I don't see many reasons to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326, #8328)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!!
- Change doc for a sip_pvt setting
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer);
This places the name of the destructor in one place only (where it
should be), eliminates the chance of errors in case you specify the wrong
destructor, and also lets the compiler do type checking on the argument,
again helping with keeping the code clean.
Same for users.
remove two duplicate definitions.
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I think this module doesn't compile, anyways, because
it has not been updated to the new module interface.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
e.g. in the Manager interface. This information is available as
a callerid (or something like that) during a call, but not when a
device is registered but silent.
It may be useful to have it available e.g. when developing a user
interface/operator panel, to map numbers to names.
experimental, so not committed to 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
