Hi,
After some testing on the latest release, i have some problem doing the following on LINUX :
Scenario : - SIP Phones behind a NAT - SER server under linux with rtpproxy launched - a 3660 cisco gateway with PSTN connectivity enabled.
When i call with SIP phone a PSTN number, everything is OK BUT no sound anywhere.
I could not find a sample ser.cfg script that reflect this scenario. Could someone send me this scenario ?
Maybe i missunderstood some things. In particular, do i need to launch two instances of ser (one for outbound proxy, another for request. If yes, how to do that)
Thanks.