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e276789669debb08d3d8a9a6af647e43e414fdc3
asterisk/channels/sip
History
Jason Parker f9834fd128 Add support for Digium Phones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8-digiumphones@357459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28 21:52:48 +00:00
..
include
Add support for Digium Phones.
2012-02-28 21:52:48 +00:00
config_parser.c
Set port to a default sane value if a bogus one is provided when parsing hostnames.
2012-01-13 20:29:03 +00:00
dialplan_functions.c
Returns with an error notice if CHANNEL function of SIP channel is read without arguments.
2011-03-08 20:19:32 +00:00
reqresp_parser.c
Misc minor fixes in reqresp_parser.c and chan_sip.c.
2012-01-19 23:17:31 +00:00
sdp_crypto.c
Allow SRTP policies to be reloaded
2012-02-24 15:07:09 +00:00
srtp.c
Add SRTP support for Asterisk
2010-06-08 05:29:08 +00:00
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