SPA3102 Line 1 & PSTN Share Same VSP A/C

Unanswered Question
Feb 25th, 2010

According to the document:

Routing calls between PSTN and VoIP with a SPA3102

Looks to me both Line 1 and PSTN Tab are share the same ITSP VoIP A/C

In the example of VoIP-To-PSTN Gateway, when caller call ITSP DID number, how can we know it will connect to Line 1 or PSTN side?  If caller being connected to Line 1 I would expect the PSTN gateway won't works because it will just ring the analog phone. Am I missed something?

Another question is that what if one stage dialing in the VoIP-To-PSTN Gateway set to "Yes", can I simply pass a PSTN no. in the SIP URI and get it dial out directly? If the answer is possitive, what would be the format?

I have this problem too.
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Alberto Montilla Mon, 03/01/2010 - 02:43

Dear Sir;

PSTN side is 5061 port and Line 1 is 5060. This is the way to route the call to one or another.

Your understanding is right about dialing. If you want to have further details, please check the ATA Admin Guide.

Regards;
Alberto

samsonluk Mon, 03/01/2010 - 03:22

Hi Alberto,

I already read page 93 of the ATA Admin Guide, it doesn't provide further details on how to make a One-Stage Dialing. It only explains what is a One-Stage Dialing and mentioned HTTP Digest Authentication can be used but doesn't provide any useful example such as the correct sip uri format for making a OSD call and page 213 also doesn't provide any details on how to construct a  HTTP Digest Authentication, etc. So I have to ask for further details here or you can point me to the right document.

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