[Asterisk-Users] [PLEASE RESPOND] Supervised transfer over SIP to outside POTS lines

[Damon Brown]

PLEASE RESPOND IF THERE'S A SOLUTION
I am trying to complete my dial plan and have come up with an
interesting situation.  My configuration is set up with 12 xlite SIP
clients on SUSE linux workstation.  They are calling out via 10 analog
lines, TE110P->rhino 24 fxo.  

It all works and dials out great ... but ... this unit was brought in to
handle the "global" office.  So the help desk support on the Suse
machines need to transfer a call to an available local rep in another
state.  I thought this was possible .... until I realized the "transfer"
only works on xPRO, which isn't available for linux.

So I cant rely on SIP to handle this, I set up my extensions.conf have
transfers, ie:         

[sip-exten]
exten => 1001,1,Dial(SIP/1001,20,Trt)
exten => 1001,2,Hangup

And features.conf is :

[featuremap]
blindxfer => *1                ; Blind transfer
;disconnect => *0               ; Disconnect
;automon => *1                  ; One Touch Record
atxfer => *2                   ; Attended transfer


OK each analog phone has three way calling on it ... can I set up a
"flash" command?  How would that be done???.  

Thanks so much!!
D