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."
~B
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!
Cheers,
~Brian
--
----------------
Brian A. Henning
strutmasters.com
336.597.2397x238
----------------
--
TriLUG mailing list : http://www.trilug.org/mailman/listinfo/trilug
TriLUG Organizational FAQ : http://trilug.org/faq/
TriLUG Member Services FAQ : http://members.trilug.org/services_faq/
