Hi everyone.
Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the dial plan for extension 1 takes care of the rest.
This works when I make a typical SIP to SIP call, but it doesn't when
I call from the console, using chan_alsa. I can see that the transfer
feature is inactive:
rasterisk*CLI> core show channeltype console
-- Info about channel driver: Console --
Device State: no
Indication: yes
Transfer : no
Capabilities: 0x40 (slin)
Digit Begin: no
Digit End: yes
Send HTML : no
Image Support: no
Text Support: yes
However, I am unable to find a way to activate it. How can I transfer
placed from the console? Is it possible, in principle?
Alex
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
