Hi there, I am new to the forum, so maybe I am posting this were I shouldn't, so please let me know if this topic does not belong here.
I have a network like the one in the picture (link here
http://i59.tinypic.com/v7e1s3.png as I couldn't upload to the post), with the following configured:
- I have created users in A2billing and use their VoIP credentials in the ATAs, which successfully register with A2billing using Asterisk Realtime
- The asterisk in the A2billing server registers with the provider
- The Rate Engine in A2billing works as the simulator provides rate for given destinations
- The calls from the users get registered in A2Billing.
- The calls from the analogue phones go through as the end phone rings, and get registered in A2Billing for the given user (although as Cancelled)
- The calls appear registered in the provider's log and are billed according to its duration.
However, no sound is heard in any of the phones after picking up, and a2billing does not recognise any duration for the call. As said in the log they appear as CANCEL and 00:00 duration.
I have tried to reduce the complexity of the problem and set up a SIP client on a mobile phone, which registers to the asterisk in the A2Billing server. Although this client does not register to A2Billing (I don't know how to do so, :-S), the calls from it experience the same behaviour as above.
Googling about the problem, I have learnt that it is common problem related to a natted networks. I have tried including externip and localnet within the [general] context of the sip.conf of the asterisk in the A2Billing server, but this produces the asterisk lose registration from the SIP provider.
I know this question is more related to asterisk than to a2billing, but I was hoping that your experience could shed some light in this regard.
Best,
carlos