Well some of the details will depend upon the site you are trying to reach. Basically you want to determine the caller is part of your domain, authenticate them if you so choose the determine how to process the INVITE message. One possible approach is:

if (uri=~"^sip:[3468][0-9]{4}@bigcompany.com") { xlog("L_INFO", "\n[SER]: Outbound on-campus call. Going to route block #2\n"); route(2); break; };

This will check the request uri to see if it begins with sip: followed by either a 3,4,6 or 8 followed by four digits with each digit ranging from 0-9. If there is a match call flow jumps to route block #2 (my own choice of numbering). Route block #2 does:

route[2] {

xlog("L_INFO", "\n[SER]: Route block #2 campus 5-digit extensions: Time: [%Tf] Method: <%rm> From uri <%fu> To < %tu> IP source address <%is> R-uri: <%ru> Contact Header: <%ct> \n\n");

rewritehostport("18.9.6.8:5060"); t_on_failure("1"); t_relay(); break; }

This is a scaled down version of what I actually do. It is for illustration purposes only. The IP address in the rewritehostport statement is the address of my primary PSTN gateway. The t_on_failure will cause call flow to be directed at the backup PSTN gateway should the primary be unavailable.

Good luck, Steve

Edgardo O. Gonzales II wrote:

Hi!

Can somebody teach me how to enable call routing .. Meaning if the called number is not on my routing table, I will forward the request to another sip gatekeeper ..

Need it so badly.. please help ..

thanks, ed

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