revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines
Remove duplicate increment of the header count in the add_header() function.
(closes issue #11648)
Reported by: makoto
Patch provided by sergee, committed patch by me, inspired by comments from putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines
Issue 11574: Add dependencies on res_monitor and res_features.
I wonder if Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends on it.
Reported by: caio1982
(closes issue #11574)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10690)
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r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines
If a typo is found in a config file, we previous continued on with what was already loaded.
We do not want to do this (see bug below for details).
This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.
Issue 10690.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 lines
Avoid reinvite race situations with two Asterisks trying
to reinvite each other in 1.4 and trunk.
This patch implements support for the 491 error code that
Asterisk 1.4 generates on situations where we get an
incoming INVITE and already has one in progress.
Thanks to mavetju for reporting and to Raj Jain for an
excellent explanation of the problem.
Patch by myself. Tested with 8 Asterisk servers connected
to each other in a training network.
Closes issue #10481
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
make the format explicit in a debug message;
print the audio instead of aggregated peer capability in a debugging msg.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
with the SIPPEER() argument of the same name. The deprecation procedure is not
being used here since this is a trunk-only option.
(closes issue #11307, reported by pj, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this, you can control the peer in the dialplan, so you avoid placing outbound
calls when the device has reached busy-level.
Reported by pj.
Closes bug #11180
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Tilghman Lesher
Joshua Colp
Olle Johansson
Luigi Rizzo
Jason Parker
Steve Murphy
Mark Michelson