[Spce-user] SIP trunk

[Jon Bonilla (Manwe)]

El Mon, 25 Jun 2012 09:24:33 +0000
Sylvester Nielsen <sn at fest.dk> escribió:

> Hi,
> 
> I have been trying to get an account to work as a SIP trunk for a VoipNow
> PBX, but I can't get it to work.
> 
> I tried to search for it and found another thread with the same issue, but no
> solution.
> http://lists.sipwise.com/pipermail/spce-user/2011-December/000926.html
> 
> I also found another asterisk as subscriber question, but i already did try
> to set those options.
> http://lists.sipwise.com/pipermail/spce-user/2012-January/000934.html
> 
> I have set e164_to_ruri and allowed all CLI's in the subscriber settings.
> 
> I made a ngrep trace of an outbound call and an inbound which is attached.
> The voip test number is +4578788471 and the number I try to call is 80808080
> 
> The inbound call looks like it was answered on the mobile phone, but the IP
> phone did not ring and no RTP was send.
> 
> Just for info, then it works fine with calls from a regular SIP phone both
> ways.
> 
> Best regards
> Sylvester Nielsen
> 
> 


Hi sylvester. 

It would be great it you could attach a network diagram to show which IP is a
subcriber, which the sip:provider and which a peer.

In the in.txt a normal call is shown. It's answered by the subscriber and a bue
is sent from the caller after some seconds. I don't know what "answered by the
mobile" means there as I see an asterisk server on the callee part.

In the out.txt I see a call sent to the spce sent (from a peer?) and the ngcp
does not find any user with that number. 

If you could please explain better the scenario and attach the kamailio-proxy
log for those calls, at least the out.txt we could help you.