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Unread 04-29-2006, 11:09 PM   #1
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Default SIP to SIP Calls and

A quick question to see if my understanding is correct. I have put my home PSTN number on e164, so presumably if anyone makes a call to my PSTN via VoIP, the call will be free if their provider looks the number up on e164 first.

The process, as I understand it, is that my home pstn 9452XXXX would get translated to my sipme number 1777XXXXXX and the call would be placed directly to the 1777XXXXXX number for a cost of $0 (in most cases depending on their VSP provider).

I've tried to test my home number with voxalot (which I thought did the e164 lookup) and was expected it to route the call directly to but instead it routed it to

Is this correct?

Another thing I don't quite understand. From reading about sipbroker, and given that I have a DrayTek 2100V, it appears that I have to route my calls to sipbroker first so that it can sort out where to send the outbound call. But since I have a DID provided by sipme, doesn't that mean that sipme would be unable to route calls to my DID (especially from the PSTN network)? So if someone phoned my DID from the PSTN, there is no way SIPME could route the call to me unless they too used sipbroker on the inbound call? So,

Call to 02XXXXXXXX (my DID) from someone on the PSTN
PSTN routes call to SIPME exchange
SIPME looks up 02XXXXXXXX on sipbroker and finds there is a match to
SIPME routes call to
***but since my ATA is not registered with sipme but with sipbroker, then sipme thinks I'm off-line and fails the call???

The only way that I can see this all working is for the VSP (sipme in my case) to use sipbroker in their routing process.

Am I on the right track or completely in left field?

Last edited by pmerrill; 04-29-2006 at 11:46 PM.
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