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mirror of https://github.com/asterisk/asterisk.git synced 2026-07-27 16:07:15 -07:00
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54d8fd09688dacedfedfcaf1ea300bfa8edcdbd2
asterisk/funcs/func_channel.c
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Joshua Colp 1961b57705 Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-26 23:31:23 +00:00

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