Hello all, I have a problem that is giving me headaches for quite a while. I have an Asterisk server running at one location, registering to different SIP providers. The server has a routable IP address, doesn't use NAT. In a different location I have a Cisco router with VoIP capabilities, connected to the PSTN through a PRI. The Cisco router also has a routable IP address. In my Asterisk server, I am using the Cisco for connecting to the PSTN. Basically I want all inbound calls from different SIP providers forwarded through the Cisco router to the PSTN and then my swll phone. Now here's the problem: let's say my Asterisk is connected to SIP provider X. If I call my account from another account of the SIP provider X, everything works OK, I get the call forwarded to my cell and the sound works both ways. But it somebody calls my account at SIP provider X from a regular PSTN line, at least one of the ends cannot hear the other (or maybe doesn't work both ways). If instead of forwarding the calls through my Cisco router I send them to an internal extension where I use a Sipura adapter, the sound is OK in both cases. Any idea where to start troubleshooting this ?
Thanks, Liviu
