[Asterisk-Users] SIP Trunking

[C F]

It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.

On 6/2/06, Steven Haldeman <haldeman_79 at yahoo.com> wrote:
>
> Hello,
>
> I am attempting to figure out how to set up SIP trunking, between my company
> and our SIP provider.  This is an expermintal project at this time.  The SIP
> provider gave us a Signalling IP address and two Media IP addresses.  We
> supplied them with the IP address of our Asterisk box.  When asked what our
> Usernames and Passwords would be we were told that they were not needed for
> a SIP trunk.  We can use what they call SIP lines that use username/password
> however because of tarrifing the lines cost more per month than a trunk I
> have been successfull in making a SIP Line work, but have no idea where to
> start with a SIP Trunk.  We sill be using DID numbers on the SIP trunks.
> Has anyone had any experiance with this type of configuration an example
> would be extrememly helpfull.
>
> I have search the internet for help and I may have seen a solution but just
> was not certain what I was looking at, or how to implement as everything
> that I have seen user a Username/Password combo.
>
> Thank you in advance.
>
> Thanks
>
> Steven
>
>
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