Siqhamo Sifo wrote:> My asterisk is giving me problems when I use it as a pstn gateway to SER ,
> basically what happens is that its either I get one way audio or no audio
> at all when I make pstn calls via asterisk from sip clients registered
> with SER.
SER itself is just a SIP Proxy. So your issue may be the fact that you
are not re-writing the SIP headers, if your endpoints are behind NAT.
Diagnose the situation then provide detailed information, if you expect
any assistance - We cannot read minds, yet.
Jeremy McNamara