[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[Olle E. Johansson]

Andres wrote:

> On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
> 
>>John:
>>are you aware of any documentation on how to configre SER to be a front-end
>>to Asterisk?

> At TeleSIP we run a cluster of several geographically distributed SER Servers 
> that hande all our SIP Routing.   SER is a robust, fast and stable platform 
> which has worked flawlessly for us.  We use * as our company PBX and PSTN 
> Gateway.  Basically what you need to do is to device a numbering plan so that 
> SERs routing logic can forward the call to * when it needs to.

Thank you for the good example!
Another example can be found in the SER handbook, found on IPtel.org. The example
mentions how to use SER as a frontend to a Cisco PSTN gateway, but also applies
to using SER as a frontend to Asterisk.

/O