Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just after sending the OK with the proper SDP to the remote Proxy, the Asterisk initiates a new INVITE to the local extension and remote Proxy, with the normal audio codecs again.
We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the local extension and remote Proxy, but it still forces the call to go back to a voice call. Any idea why it happens and how to debug it? We set verbose and debug to 20, but no "internal" info is provided to get a clear understanding on Asterisk's "thoughts" during that process. Thank you in advance for your assistance, Andreas _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users