One way audio att uverse

General Help
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I’m still waiting for an actual debug that shows the call that has audio issues. The last one was just a bunch of OPTIONS packets which are not a call.

Is 0027 the extension that is having issue? Is this the remote extension?

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On RTP port forwarding:

What you do with the SIP Signaling port has almost nothing to do with your audio. Audio transits your network on a pair of address typically randomly designated in the 10000-20000 UDP port range. This is why you port forward UDP 10000-20000 on your gateway firewall to the PBX server. Once again, this is different than the signaling port (5060, 5160, etc.) for the SIP part of the connection.

The other thing that can hose you up is that SIP includes NAT forwarding information in the signaling packets. This is why it’s important for your system to have both the internal address of the server (inside the firewall) and exterior (routable firewall) address. The SIP protocol includes all of the interior and exterior addresses so that the firewall can be successfully traversed.

One-way audio is almost always a problem with NAT. Make sure that the remote extension is set up to perform NAT correctly and make sure that the extension definition in the Asterisk server is set correctly. Also, double check your Advanced SIP settings in the Advanced tab to make sure that all of the internal and external addresses are set up correctly.

If you are using the Adaptive Firewall to pass this guy in, you shouldn’t “need” for port forward 10000-20000 from the firewall to the server, but adding that is the first thing I’d try. The problem may just be as simple as the audio is getting dropped at your firewall because there’s nothing listening for the traffic where he’s sending it. Remember - you need to make sure that both ends are set up correctly and that the firewalls at both ends are passing the RTP traffic correctly from the phone to the server and back.

@aspieboy77 - to reiterate a little bit:

  1. Make sure the firewall at the remote phone’s location is set up with SIP ALG off. If you are using a VPN, skip down to 12)
  2. Set up the remote extension to use NAT and make sure the NAT settings in the phone are correct;
  3. If the remote extension’s IP address is dynamic, configure your phone using DYNDNS and a dynamic host name at both ends. If the IP address is static, skip to 5
  4. Another COA for dynamic addresses, you can use the Adaptive Firewall to allow access from the remote address. In this case, you will need to use a STUN server to get the actual routable address for the firewall so you can use it in your phone.
  5. The FreePBX Firewall should be configured to access traffic from the DYNDNS hostname or the STATIC IP address.
  6. Block out all of the IP addresses’ access to port 5060 and 5160 to only your ITSP.
  7. Add in the STATIC IP address for your phone in the firewall.
  8. In your gateway firewall, allow access to port 5060 and 5160 and any static IP addresses you have phone coming through. If you have dynamic addresses, you should be able to get the possible routable ranges from your “remote” provider. If you NEED to open this up to the world, you need to otherwise you need to NOT open this up to the world.
  9. Port forward UDP/5060 and UDP/5160 to your PBX. The FreePBX Firewall will do the rest of the filtering.
  10. Port forward UDP/10000-20000 to your PBX. This is generally safe as long as you don’t have servers listening on these ports on other servers.
  11. Back up your configuration in case there’s a problem, and document what you did on the phone so you don’t have to remember all the details. Stop here.
  12. If you are using a VPN, you do everything in the FreePBX Firewall. Set up all of the addresses in the VPN as “local” addresses and you should be good to go. You should be able to skip all of this junk and go straight to making phone calls.

With all of those “features” set up, you should be in pretty good shape.

1 Like
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yes 0027 is the remote extension that is having issues and i thougt i gave you the log if not ill have to do some more digging i’m sorry to be such a pain i just dont know linux that well

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ok i will get on this tonight

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Welll if this is the remote extension you already have some problems. It’s not setup to tell Asterisk it’s behind NAT. You need to go into the Advanced tab and set the NAT setting to Yes. That is step 1.

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