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- implement all functions through internal APIs in res_osp.c and app_osplookup.c
(homesick)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18369 f38db490-d61c-443f-a65b-d21fe96a405b
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application to partially
demonstrate the capability of manager over http.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16850 f38db490-d61c-443f-a65b-d21fe96a405b
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It's nice to have consistency in sample configs too.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16268 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15518 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #6390)
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modifications)
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- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15148 f38db490-d61c-443f-a65b-d21fe96a405b
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- Implement option for allow/disallow subscriptions
- Implement option for allow/disallow overlap dialling
- Set default to disable overlap dialling in sip.conf.sample for new installations
- Remove overlap dialling from subscription logic
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on reload.
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add a WAY cool new field to the nsf option
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14521 f38db490-d61c-443f-a65b-d21fe96a405b
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international inbound calls.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14079 f38db490-d61c-443f-a65b-d21fe96a405b
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the Asterisk file format and codec translator modules (issue #6062)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r13964 | russell | 2006-03-21 13:59:29 -0500 (Tue, 21 Mar 2006) | 3 lines
add a note explaining how to set the DYNAMIC_FEATURES variable to allow the use
of custom features (issue #6747)
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config option hdlc=yes now which enables the hdlc controller for a data call
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that you're required to read on... (hello xrobau)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13246 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #6665)
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the "h" extension as opposed to afterwards (issue #6193)
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cdr_tds.conf. patch provided by bug 6629 with
minor change.
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changed the default from 1 to 2000. The default for the upper_threshold is now 0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12287 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r11946 | russell | 2006-03-05 21:32:35 -0500 (Sun, 05 Mar 2006) | 2 lines
fix a typo in the description of the ringtimeout option
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11947 f38db490-d61c-443f-a65b-d21fe96a405b
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usegmtime, log date/time in GMT
loguniqueid log uniqueid
loguserfield log user field
patch provided by bug 5015
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11192 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r10736 | kpfleming | 2006-02-22 11:32:15 -0600 (Wed, 22 Feb 2006) | 2 lines
add comment warning people about trying to use hostnames/IPs in the sample config
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config to other asterisk samples , bug note 6530
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10559 f38db490-d61c-443f-a65b-d21fe96a405b
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remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
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#6315, plus documentation)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10163 f38db490-d61c-443f-a65b-d21fe96a405b
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asterisk standard, changed it so it can take frames
of sizes other than 20ms, allowed for the app to reload
properly, and finalyl changed sample to general section
as to follow standards.
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fixes. Thanks to all those involved.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9783 f38db490-d61c-443f-a65b-d21fe96a405b
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