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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
