Olle Johansson 404151ad65 New sip.conf option for setting default tonezone for channel or individual devices ...

Review: https://reviewboard.asterisk.org/r/1429/

(closes issue ASTERISK-18497)

Thanks to russellb for peer review.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2011-09-12 13:57:57 +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

Revert early destruction of RTP sessions

2010-07-13 21:42:42 +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

Merged revisions 304245 via svnmerge from

2011-01-26 20:44:47 +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

New sip.conf option for setting default tonezone for channel or individual devices

2011-09-12 13:57:57 +00:00

srtp.h

Add SRTP support for Asterisk

2010-06-08 05:29:08 +00:00