search for: mustreinvite

Displaying 1 result from an estimated 1 matches for "mustreinvite".

2007 Apr 27
1
SIP<->H323 calls without proxying RTP
Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone