[Asterisk-Users] Asterisk - SER Configuration

[Matt Riddell]

Alberto Zuin wrote:
> 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 user at a.com the request travels to 
Asterisk which dials a number on the SER box (user at a.com).

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 
number at sineapps.com 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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