is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to
transmit_reinvite_with_sdp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
lock when needed - when we remove the dialog from the dialog list
If this doesn't lead to severe problems, it might help with some locking issues
in 1.4/1.2.
- Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no
one understands it... ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines
Instead of iterating all of the options once to look for jitterbuffer options,
and then again for everything else, move the processing of jitterbuffer
options into the main loop so that there are no erroneous messages about
ignoring unknown options. (issue #8226)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
appreciated really. (Read the coding guidelines).
I've worked hard to make chan_sip a better place to code in, let's
keep it that way and don't add more stuff without comments.
Thank you.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
