Yes, but I have to configure a route for each host in every host! A the moment i have about 120 Asterisk hosts and every astersk have about 50-100 users! Is for that I want a single sip proxy that route dial. I read more about ser, and the suggestion is to use ser for accounting and route, and asterisk only for PBX gateway and for voicemail. In my situation this isn't perfect because I have to use asterisk for sip login...
What you do in this situation:
Remember the point that Asterisk is a UA, not a proxy. You get Asterisk to register to SER with a particular account.
When one of the other boxes dials [EMAIL PROTECTED] the request travels to Asterisk which dials a number on the SER box ([EMAIL PROTECTED]).
SER looks in it's routing table to see where a.com is, and redirects the request there.
Once the request gets to the Asterisk box at a.com, the Asterisk server checks the account name that the request is for and forwards it to the user. With record routing obviously the 100 - Trying, 180 - Ringing and 200 ok pass through all of the previous servers.
This allows you to keep control of accounting etc at any box along the way. (I.E. one of your rules in SER might say that if a call is to a [EMAIL PROTECTED] then pass it to a PSTN gateway). With the record routing on, you would still get a message saying that the call had hung up even if you are not one end of the call.
Your best bet would be to read up on some of the SIP documentation on the iptel.org site (particularly the introduction to SIP and the SER user's guide).
Hope my ramblings make sense!
:)
-- Cheers,
Matt Riddell _______________________________________________
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