Subject: [Sip] SIP - PSTN Interconnection

From: "Jagannathan R" <[email-masked]>
Date: Thu, 21 Feb 2002 12:07:27 +0530

I have a some basic doubts regarding the SIP-PSTN Interworking. Can any one clarify these doubts please.

1. Proxy server is connected to MGC (point of SIP-PSTN Interconnection), is there a way proxy server registering itself with MGC. This is required at MGC to know what are the SIP phone numbers this proxy can service, so that the PSTN originated call can be correctly terminated to the Proxy. 
1b. And also the MGC should know the status of Proxy (In-Service or Out-of-Service) to terminate the call.
1c. How many Proxy servers with different/same set of SIP phone numbers can be linked to MGC.

2. When SIP call arrives at MGC, MGC requires NOA (Nature of Address) in order to translate the called number and route the call to the correct PSTN Trunk Group. How the NOA address (more specifically + 0, - 0, 0+IDDD, IDDD, DDD calls) can be represented in the SIP messages.

3. For SIP-PSTN call, how different digit sequences like ANI, Authorization code, Account code (these are required for billing at MGC and to have Dialing plan) can be send to MGC in SIP Messages. 

Thanks in advance,

jagannathan