Issues with Asterisk and MPs behind NAT(s)

[Carlos Rey-Moreno]

Good afternoon,  

We have managed to set up asterisk/a2billing in a network like the one in the picture (http://i59.tinypic.com/v7e1s3.png), with the following working: 

- The asterisk in the A2billing server registers with the provider

- The calls from the analogue phones go through asterisk/a2billing 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 hope someone can help us with this

Best, 

carlos

--

Carlos Rey-Moreno
Research Assistant
Office 1.28
Department of Computer Science
University of the Western Cape
Private Bag X17 - Bellville, 7535
Cape Town - South Africa
Tel: +27 (0) 21 959 2562 Cel: +27 (0) 76 986 3633
Skype: carlos.reymoreno Twitter: Creym

[Wayne Abroue]

Hi Carlos

I solved the problem by using IAX connection to the provider. Who are u using?

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[Carlos Rey-Moreno]

Hi there, 

I managed to get hold of a USB dongle (obtaining a public IP) and set it up on the a2billing server. Then in each asterisk in the Mesh Potato cloud I included localnet=10.130.1.0/255.255.255.255.0 and externip=0.0.0.0 (uncommenting the lines in potato.sip.conf) and it worked. Now the audio works bidirectionally, and the calls are billed in a2billing.

We will submit a HowTo soon.

best, 

carlos

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