[TriLUG] Asterisk: SIP call transfer
brian at strutmasters.com
Tue Jun 6 14:36:08 EDT 2006
Ha! Nevermind me. :-) I found it.
For the curious: The Dial(...) which originates the call to the SIP
client has to include the "t" option ( Dial(SIP/userid,timeout,options)
), then if the SIP client dials a #, Asterisk jumps in and pushes the
call back into the dialplan on whatever extension the SIP client dials
within the client's context. It's the "t" option that tells Asterisk,
"Allow the recipient to transfer the call."
Brian Henning wrote:
> Howdy gang!
> I'm proud to say I have a good 90% of the functionality I'm after up
> and running with my brand new Asterisk setup. The 10% I'm missing is
> how to let a SIP user transfer a call back into the dialplan (wherein I
> can work some magic to get our PBX to then transfer said call to a
> selected extension). The call to be transferred will always originate
> from our Nortel PBX and be piped into the Asterisk server via a Nortel
> ATA and a TDM400P FXO module. So it's really just a matter of being
> able to tell Asterisk to flash the FXO line and dial the transfer code
> (easy) from the middle of an established SIP call (not so easy, so far).
> The SIP client is X-Lite. I tried Gizmo, but couldn't seem to get it
> to register with the * server.
> Many many thanks to all the folks who've lent helpful info so far, and
> thanks in advance to those who will continue (or start!) to do so!
Brian A. Henning
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