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94e4733d89e9616f27888d0a88efc135318ecab4
asterisk/channels/sip/include
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Matthew Jordan 1cb1d04679 Whitespace cleanup
Remove some extraneous blobs


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-19 14:53:49 +00:00
..
config_parser.h
Stasis: Update security events to use Stasis
2013-05-17 17:36:10 +00:00
dialog.h
logger: Adds additional support for call id logging and chan_sip specific stuff
2012-05-17 16:28:20 +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
Asterisk media architecture conversion - no more format bitfields
2011-02-03 16:22:10 +00:00
reqresp_parser.h
Fix Record-Route parsing for large headers.
2013-01-18 05:31:23 +00:00
security_events.h
Merged revisions 337595,337597 via svnmerge from
2011-09-22 16:35:20 +00:00
sip_utils.h
Add auto_force_rport and auto_comedia NAT options
2012-02-09 18:14:39 +00:00
sip.h
Whitespace cleanup
2013-08-19 14:53:49 +00:00
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