[asterisk-users] SIP trunk

[Tim St. Pierre]

Make a context called DID or something like that, and set your peer entry in 
sip.conf to have your provider's calls go tho this context.  The incoming SIP 
invites will be directed to the DID number at your server.

Use Goto to direct the calls where you want them to end up.

ie.
[DID]
exten => 6477226929,1,Goto(phones|5101|1)
exten => 6477226930,1,Goto(ea-mainmenu|s|1)


On September 11, 2006 07:30, Richard Klingler wrote:
> hello
>
>
> If I want to use asterisk to hookup to a SIP account
> I just use the "register" line in sip.conf with the
> extension number at the end...
>
>
> But how about if I want to use a SIP trunk from a
> provider which gives me 10 DID numbers with the same account?
>
>
> thanx in advance
> rick
>
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-- 
Tim St. Pierre

IP telephony specialist
sip://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net
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