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

[brett-asterisk at worldcall.net]

Kevin P. Fleming wrote:

> brett-asterisk at worldcall.net wrote:
>
>> businesses, it's no problem. BTW, implementing LNP with a SS7 
>> connected switch is really quite easy. Your provider shouldn't be 
>> giving you any 
>
>
> "really quite easy"? That may be true if you are _already_ a carrier 
> with OCN(s) and LRN(s), but getting to that point is not easy in many 
> places in this country.
>
> What sort of switch are you using do to the TDM<->IP conversion, since 
> you say you are using SS7 (which Asterisk does not yet deal with) and 
> that you see LRNs in your SIP headers (which means the TDM<->IP 
> conversion is happening outside your Asterisk box)? Presumably you 
> have some sort of Class 4 switch that can route LRNs to a SIP 
> destination, or you wouldn't be able to direct numbers to your 
> Asterisk boxes transparently...


Yes.. if you are already a carrier it is easy. If you arn't...... it's a 
long road.. It took me.. maybe 6 months to get it all set up..

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

-Brett