Asterisk PBX call transmit audio stops working after about 5 minutes

The call connects just fine and after about 5 minutes (sometimes less) it will stop tranmitting your audio. You can receive audio just fine. The trunk is from VIAtalk.com
mspurlingAsked:
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grbladesConnect With a Mentor Commented:
Its the rtp audio UDP ports 8766 to 35000 which need to be forwarded. You can specify a much smaller port range on the device first if you wish as it doesnt need to be that big. 10 ports should be sufficient.

You shouldnt need to worry about forwarded the SIP ports. Just set the registry expiry time small (5 minutes for example) so that the firewall doesnt close the connection thinking it is inactive.

I wouldnt bother trying to use a tunnel directly between the equipment. Its too complex.
You could use a hardware firewall such as a Cisco ASA5505 and smaller devices or a cisco router at the other sites and have a VPN connection but that should really be as a last resort.
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grbladesCommented:
Is there any NAT device between the asterisk box and the phone or viatalk?
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mspurlingAuthor Commented:
I have tried on both sides of the Firewall and it does it. They are Linksys pap2t-na adapters.
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grbladesCommented:
What firewall do you have?
What ports have you forwarded through on the firewall so far?
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mspurlingAuthor Commented:
Ok, the PBX side is not behind a firewall. It is on the public side of the internet completely open (won't be permantly but now it is) The Linksys devices have also been on the public side on the internet (same switch no hops) and behind firewalls (two hosp away) but have not opened any ports on those firewalls. Do I need to since it works sometimes?
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grbladesCommented:
Ok I'll assume asterisk is connected directly to the internet with no firewall inbetween.
If the linksys devices are behind a firewall then you should really forward the rtp ports configured on the linksys. Since different ports are used in each direction the firewall can think the traffic has stopped since their are no replies and close the connection. Forwarding the ports signals that new udp connections are permitted so it wont look for replies.

10pm here in the uk now. I will reply to any furthur posts tomorrow.
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mspurlingAuthor Commented:
Ok, what are the RTP port ranges that need to be forwarded? Are they?
SIP signaling: Ports 5060 to 5070
RTP audio: Ports 8766 to 35000
What do you think about an IP Tunnel? I think I read somewhere that you can do that between the linksys terminal adapter and directly with Asterisk but I can't seem to find that now.
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mspurlingAuthor Commented:
I adjusted this and it didn't seem to make a difference. Any other ideas? How does one setup a tunnel with in the terminal adapter and the pbx?
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grbladesCommented:
No sorry no other ideas. I wouldnt try and use a software VPN on the asterisk box. Far too many issues with NAT and firewalls that unless you know exactly what you are doing and where each layer sits in the networking stack you will probably cause more problems.
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