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Unread 10-17-2008, 04:42 AM   #6
olegp
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Hi Emoci, DracoFelis, thank you for responses.
Regarding Packet8 – they do not seem to be open to any interfacing. There are some old success stories in forums, but nothing seems to be working now. Although I will check FWD again.

I tried to use # at the end of dialed number. It did not help. BTW, I've discovered that some SipBroker PSTN numbers work (with or without #), while other do not work (regardless of #) when calling from Packet8 phone.

Speaking about Re-Invite… I can’t believe that it may be the case. As I already wrote my ATA does not receive any SIP request at all. From ATA point of view Re-Invite begins from normal "Invite" request.

Packet8>> Packet8 Termination Provider>> PSTN>>Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device
IMHO the feasibility to streamline call from Packet8 via PSTN via SipBroker to User's ATA is doubtful. To do it SipBroker needs knowledge about Packet8 VOIP side – which I guess may NOT come through PSTN link. Do I miss something here?
I compared SIP sessions for two calls (using SipBroker PSTN number which worked): from cell phone and from Packet8 phone.
- in both cases original SIP Invite comes from SipBroker’s IP (64.34.162.221)
- both originally have the same RTP source IP (sipbroker1.telengy.net)
- both re-invited to the same RTP (66.193.176.50)

So? Where we are? Still experimenting… Returning to my original question (forward did not work) – I found out that PAP2T can receive direct IP calls on the same line which already registered with a provider (voipstunt.com) – "Ans Call Without Reg" in PAP2T settings should be changed to "yes". Second condition - UserID should match, but did not in my case. As prove of concept I have coded simple UDP proxy (a program running in PC) – accepting incoming request, replacing UserID and forwarding request to PAP2T – worked like a charm! (As usual there is huge gap between 10-20 lines of code as prove of concept and real program which I have no time to finish). Also I discovered MySipSwitch.com – it seems to employ the same idea…

Once again, thank you guys for the support! VOIP is really exciting area!
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