[asterisk-users] Sip Trunking

[Benny Amorsen]

Brent Davidson <brent at texascountrytitle.com> writes:

> I have several branch offices, each with their own Asterisk server 
> (version 1.4.22.1) handling their PBX functions.  All of these offices 
> need to talk to each other.  In sip.conf I created a peer entry for each 
> office with a username of branch-user and a friend entry for every 
> branch-user with the username being just the branch, for example:

You should only need peer entries... type=user is dying.


/Benny