Andre: This makes a lot of sense. I had used Asterisk in the past to play the role of Gatekeeper for directing traffic to the appropriate Asterisk acting as a PSTN gateway. IAX does a heck of a good job in that configuration. However, with SIP, I have run into nothing but trouble with registrations falling off. I have read the SER manual I am going to jump into it, now that I know that in "practice" it works and it is not only theory in a manual. Thank you, Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andres Sent: Tuesday, October 14, 2003 12:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) 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? Hi Uriel, 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. For example in ser.cfg you could put something like this: ############################################# ### PSTN ACCESS ####### ############################################# if (method=="INVITE") { if (uri=~"sip:[EMAIL PROTECTED]") { log(1, "This is a Long Distance Call\n"); route(6); break; }; }; _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users