[Asterisk-biz] LiveVoip Has Level 3 DID's

[Kevin P. Fleming]

brett-asterisk at worldcall.net wrote:

> I am not terminating SS7 to asterisk. The current options to do so arn't 
> very practical. I currently use Sonus gear as my PSTN to SIP gateway.
> Here's what I see on a call to a ported number to my asterisk server.. I 
> think it's kinda neat:
> Assuming the customer number is 7135551212 for purposes of discussion.
> INVITE 
> sip:7135551212;rn=7132310099;npdi=yes at 192.168.100.10:5060;dtg=TRUNKGROUP01;user=phone 
> SIP/2.0

Yeah, that is kind of cool :-)