I am using Sipx 4.0.0.
I have a webpage that initiates a call between 2 psnt phones via an unmanaged gateway.
The webpages is registered with sipx extension 200. The webpage dials the first pstn number and when he answers the webpage
transfers (REFER) the call to the second pstn number. This callflow is working. The problem is, I want to know the duration the call between
the 2 pstn numbers, which is not possible with this callflow.
So what I want to do is change the transfer into an invite. So the webpage sipx extension will be in conference with the 2 pstn lines.
Is the following callflow possible with sipX ?
with A the sipx extension, B pstn number 1, C pstn number 2
1)INVITE (A2B) 2) 180 RINGING 3)200 OK 4) ACK (A2B) 5) INVITE (B2A) 6) 200 OK
7)ACK (B2A) 8) INVITE(A2C) 9) 180 RINGING 10) 200 OK 11) ACK (A2C)
Thanks !
MdM
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