[Asterisk-Users] Multiple SIP gateway accounts
Adam Greenbaum
lists at refinitive.com
Thu Oct 28 06:38:26 MST 2004
On Wed, 2004-10-27 at 15:58, Adam Greenbaum wrote:
> If you have multiple accounts on the same SIP-PSTN gateway, how do you
> dial out of a particular one? I think the answer will also involve me
> setting my domain and username on the outgoing invite, but I have a
> feeling this might not work because of the authentication.
Ok, to answer my own question, it looks as though the correct way of
doing this is to use number at ENTITYNAME (from the sip.conf) in address
you are dialing. Then fromuser and fromdomain in the SIP entity. Would
anyone comment whether this is correct?
This now brings me onto another question:
How do I associate a SIP entity with a registered account on a PSTN
gateway?
I have 2 register lines and 2 entities. When I dial into asterisk from
the PSTN gateway it always associates with the second entity. (CVS)
I've looked through the source and it _seems_ as though you can only
match against the from: username [find_user()] and from source address
[find_peer()].
Surely you would need to match against the destination sip: username,
not the From: username. Am I missing something? I must be, otherwise you
would never be able to use multiple accounts on a SIP gateway.
Thanks for your help,
Adam.
More information about the asterisk-users
mailing list