[asterisk-users] Forcing SIP trunk matching order?

[Duncan Turnbull]

Hi James

On 29/06/2012, at 6:19 AM, James Lamanna wrote:

> Hi,
> I have a bunch of different customers on an Asterisk Box (the PBX).
> This Asterisk Box is behind another Asterisk box that provides a PSTN
> connection.
> Up to this point I've been using IAX between the 2 Asterisk boxes, but
> I would like to use SIP instead.
> After doing some testing I have the following issue.
> 
> If customer A calls customer B, but the call goes out through the PSTN
> and comes back in, the call is rejected at the PBX because it wants
> authentication.

This raises the question of how calls come in from the PSTN in the first place. I am guessing you route them out in order to bill them? I am guessing you have more than one SIP trunk between the two boxes for different purposes and what you are saying is the authentication is falling to the lowest common denominator.

In this situation you could separately register two lines, raise your level of security ( insecure = no option), you need to make sure there is an incoming context that matches the supplied user name otherwise it will fall back to the ip address and whatever context that matches


> I can guess that this must be because it matches the "To" address to
> the friend SIP trunk that provides registration for the phone.
> (All phones are setup as type=friend, host=dynamic). Is there any way
> to force matching against the peer SIP trunk to PSTN so as to not
> require authentication for calls
> coming in from the PSTN server?
> 
Using permit and deny directives for the ip address should help here

> Thanks.
> 
> -- James
> 
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