michael koehler
2004-Jul-29 05:29 UTC
[Asterisk-Users] SIP and RTP / 302 after 18x / Call forwarding after announce
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@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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 935 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040729/5523452b/attachment.bin