El Sábado, 11 de Julio de 2009, Jeff Pyle escribió: > But, > the PSTN gateway --> SIP Phone audio still relays to Asterisk,
Most probably, your PSTN gateway doesn't support/allow media address change during a call, this is, it doesn't react when Asterisk sends it a re-INVITE with a new media address in the SDP and the Gw remains using the first SDP. -- Iñaki Baz Castillo <[email protected]> _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
