[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Iñaki Baz Castillo]

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 <ibc at aliax.net>