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asterisk/channels/sip/include
Matthew Jordan bc1e13dfc3
channels/chan_sip: Ensure that a BYE is sent during INVITE w/Replaces transfer
10 years ago
..
config_parser.h Add auto_force_rport and auto_comedia NAT options 13 years ago
dialog.h logger: Adds additional support for call id logging and chan_sip specific stuff 13 years ago
dialplan_functions.h Make all of the various rtpqos parameters in this branch available from the CHANNEL function. 15 years ago
globals.h Asterisk media architecture conversion - no more format bitfields 15 years ago
reqresp_parser.h channels/sip/reqresp_parser: Fix unit tests for r426594 11 years ago
sdp_crypto.h Merged revisions 336936 via svnmerge from 14 years ago
security_events.h Merged revisions 337595,337597 via svnmerge from 14 years ago
sip.h channels/chan_sip: Ensure that a BYE is sent during INVITE w/Replaces transfer 10 years ago
sip_utils.h Add auto_force_rport and auto_comedia NAT options 13 years ago
srtp.h Merged revisions 336936 via svnmerge from 14 years ago