audio problem

[horses]

I have read through the threads and don't find an answer. I have PIAF setting in a nat environment. Ports forwarded to the piaf etc. TDM400P set up, can't see trunks in FOP but that is another problem. Calls from pstn phone line work fine, audio both ways. From a soft phone, no nat there, *43 echo works. Try to call out to my cell from the softphone, answer the cell but no audio either direction.

Can't understand why calls in and the echo would work but calls out there is no audio either direction.

Pointers?

tk

 

[dobbs]

From what you've described, one stab in the dark would be to make sure you've set up your outbound zap trunks. You probably have a g0 trunk by default, but try setting up four zap trunks, just name them 1, 2, 3, and 4. Then try adding those to an outbound route and see if you have any luck.

 

[horses]

Question on doing that then. If I have four lines and they roll over on incoming calls, won't defining outbound routes pose a problem?
I had this setup on a trixbox, left them because of issues, and it worked. Only difference is the nat factor which give the results of the above I wouldn't think that is the issue.
tk

 

[dobbs]

The outbound trunks can be set to roll over as well, and Asterisk should detect when a line is busy and roll to the next one you define. Specify 1 channel per trunk in the FreePBX setting. There was a bug in that some time back, but I think it's been fixed.

If you think the issue is NAT, well that's a different story. Try reading the Nerd Vittles Orgamatron guide (even if you aren't using it) and try the various firewall advise there.

 

[horses]

k, think the first thing I will do is move it from the nat and try it there just to make sure. But wouldn't think it is a nat issue given the echo results. If it is not then will try the define trunk and roll over. Have to figure that one out since I have never had to do it.

tk

 

[jmullinix]

You might want to look at what codecs you are using. Set everything for ulaw and see if that resolves it.

 

[horses]

That may be it. I forgot to set any in the extensions. So should be disallow all and allow ulaw on each extension in freebpx correct?

 

[horses]

Tried with the extension set for ulaw. Get two way audio on first call in. Call out, get two way audio. Call in again, one way audio from outside to in. Call outs after that all have two way audio.
Any ideas from anyone?

tk

 

[horses]

Solved

Well, it was the router. Had a sip helper in there that didn't help. Turned that off and it works great. Volume sip to sip and pstn to sip seems lower than my previous though.

horses