Michelle Dupuis
2007-Feb-22 14:08 UTC
[asterisk-users] Passing call status/progress between protocols
We have a * box with sip in, and h.323 out. When the H.323 call setup is underway, will Asterisk translate the progress/status/result codes to SIP automatically? Or....do we have create our own result codes in SIP headers? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/f777b8c8/attachment.htm