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daf9607e23c5372f7794d3062b200bb456459394
asterisk/channels/sip/include
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
..
config_parser.h
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
2010-02-17 06:25:15 +00:00
dialog.h
RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer
2012-01-30 21:57:49 +00:00
dialplan_functions.h
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
2010-02-17 06:25:15 +00:00
globals.h
Add IPv6 to Asterisk.
2010-07-08 22:08:07 +00:00
reqresp_parser.h
Update reqresp_parser parse_uri doxygen comments.
2011-11-14 19:00:28 +00:00
sdp_crypto.h
Add SRTP support for Asterisk
2010-06-08 05:29:08 +00:00
sip_utils.h
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
2010-02-17 06:25:15 +00:00
sip.h
Add support for Digium Phones.
2012-02-28 21:52:48 +00:00
srtp.h
Add SRTP support for Asterisk
2010-06-08 05:29:08 +00:00
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