[asterisk-users] One way audio - doesn't seem to be NAT issue

D'Arcy J.M. Cain darcy at Vex.Net
Tue Aug 11 14:10:44 CDT 2015


I have been banging my head against the wall for weeks now on this
one.  I have a switch running NetBSD and Asterisk 11.19.0 although I
have had this problem on older versions as well.  I, and my users, can
call out, we can receive calls, quality is excellent but I cannot talk
with one user.  The different elements are as follows:

The switch as described above which is in a server room on the Internet
backbone with a public IP address.

My home system which is behind a bridged modem through a Linksys
WRT54GS with priority given to my ATA.  The ATA is a Cisco SPA112.  I
also have an actual SIP phone.  The problem happens with both.
Obviously I am using NAT but both devices work just fine if I am going
to the PSTN.

My user who is also going through a bridged modem to a Linksys SPA-2102
which is doing the PPPOE so it has a public IP address and no NAT
involved although it serves NAT for the connected computer.

So here is the problem.  While both of us have no problems externally,
when we call each other we get one way audio and it is always from me
to him no matter who initiates the call.

A further test, I can call from the SIP phone to the ATA connected
phone and vice versa just fine.  That involves two devices behind the
same NAT but since they still need to use the server as an intermediary
I can't see how that would matter.

Given that both of us can make and accept calls and the server is
simply connecting two separate channels I can't see where the problem
might lie.  Can anyone suggest a possible setup issue?

I have tried so many things but I am willing to try them again.  Feel
free to make any suggestion no matter how silly.  I really need to fix
this.

Cheers.


-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net



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