Experts asked now:
Is there a way to make this call scenario possible:
After an INVITE was received at the asterisk an announcement should be played, then, the
caller should be forwarded to another loc. REFER should not be used in any way!
I thought about something like this:
Client Asterisk
---------------------------------------------------------------
INVITE >
< 183 Session Progress
< RTP Stream
[ .. some time .. ]
< 302 Moved .. Contact: [EMAIL PROTECTED]
ACK >
But i could not figure out how to make a answer/playback happen without
the final (200 ok) response to the INVITE dialog. I thought about patching
the chan_sip, but this would take me away from the branch!?
Please only answer if:
- you know a solution (none sip REFER!)
- you may have just an idea (working or not - not important :) )
Sincerely ,
Michael
