[Asterisk-Users] SIP and RTP / 302 after 18x / Call forwarding after announce

[michael koehler]

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: otheruser at otherserver.tld
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

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