Hi,

I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 
1.6.2.  Users connect to the SIP server and dial; the SIP server 
forwards the call to the PSTN server over IAX2, which then dials out 
over the connected PRI.  Since users need detailed call progress 
feedback, the first action in the dialplan on the PSTN server side is 
Answer().

In this scenario it's easy for a human to know when a call has been 
answered.  However, the SIP-side Asterisk treats the call as answered 
the moment the PSTN server executes Answer().  Is there any way of 
determining on the SIP side when the called party actually picks up the 
phone?  Or if she doesn't, the status of the call as it progresses?

Regards,

-- Raj
-- 
Raj Mathur                r...@kandalaya.org      http://kandalaya.org/
       GPG: 78D4 FC67 367F 40E2 0DD5  0FEF C968 D0EF CC68 D17F
PsyTrance & Chill: http://schizoid.in/   ||   It is the mind that moves

--
_____________________________________________________________________
-- 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

Reply via email to