[asterisk-users] SIP & NAT ...

[Tom Rymes]

On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:

[snip]

> Both these SIP -> external PSTN provider connections register OK on  
> the * box, and outgoing calls placed over either connection works  
> perfectly. Outgoing callerId (set by the external provider) works  
> as expected. ) I have dialling prefixes for each 'line', nothing  
> special there, that side of it all works as expected.
>
> The problem is that only the last one in the sip.conf file actually  
> accepts incoming calls when dialled from the PSTN side. (They have  
> different PSTN phone numbers) If I swap their entries over in the  
> sip.conf file, then the other one takes the calls.

[snip]

I may be mistaken here, but don't you need to use different ports for  
each line? ie: Port 5060 for line 1 and 5061 for line 2?

Tom