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ben.carrasco/freeswitch
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mirror of https://github.com/signalwire/freeswitch.git synced 2026-10-18 09:14:14 +00:00
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4f39fe92aba74749fc9a5e5ac2ad473da8e5f8a3
freeswitch/src/mod/endpoints
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Brian West 4f39fe92ab sip is stupid MODSOFIA-51, the 202 has to have the correct contact or the phone comes back and subscribes to the contact and not the actual extension
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@16500 d0543943-73ff-0310-b7d9-9358b9ac24b2
2010-01-24 19:04:48 +00:00
..
mod_alsa
code before decl
2009-12-07 00:07:23 +00:00
mod_dingaling
SCA tweaks
2010-01-13 01:40:11 +00:00
mod_h323
add PI handling in Call Proceeding
2009-12-12 16:48:50 +00:00
mod_loopback
MODENDP-276
2010-01-18 17:09:05 +00:00
mod_opal
dos2unix
2009-12-15 21:25:49 +00:00
mod_portaudio
MODENDP-282
2010-01-18 04:32:15 +00:00
mod_reference
refactor enterprise_originate
2009-11-20 02:17:08 +00:00
mod_skypiax
skypiax: changed audio i/o tcp socket size and timing constant on linux, for better compatibility with virtual machines
2010-01-16 17:15:12 +00:00
mod_sofia
sip is stupid MODSOFIA-51, the 202 has to have the correct contact or the phone comes back and subscribes to the contact and not the actual extension
2010-01-24 19:04:48 +00:00
mod_unicall
refactor enterprise_originate
2009-11-20 02:17:08 +00:00
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