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

[Uriel Carrasquilla]

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: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Andres
Sent: Tuesday, October 14, 2003 12:43 AM
To: asterisk-users at lists.digium.com
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:01[1-9][0-9]+ at .*") {
                      log(1, "This is a Long Distance Call\n");
                      route(6);
                      break;
                      };
                  };