[asterisk-users] Reinvites and SIP/RTP

[Adrian Marsh]

Hi All,

 

When I use re-invite, does the Asterisk server stay in the SIP
conversation, and just RTP traffic diverts, or does the SIP transfer
away from the A*k server too ?

 

Thanks,

 

Adrian

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080715/93453030/attachment.htm