Hi Matt -

I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.

On the customer's end I have the following config in iax.conf:
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
jitterbuffer=yes
forcejitterbuffer=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1

If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and
minexcesbuffer don't do anything.  They are ignored by 1.2.x unless
you specify that you want to use the old 1.0.x jitterbuffer.  Instead
you might try the parameters maxjitterbuffer, resyncthreshold, and
maxjitterinterps.  For more, you can check out the sample iax.conf.

I believe, also, that you are correct in setting trunk=no.  I know in
the 1.0.x jitterbuffer, trunk was not fully supported.  I think this
is still the case with the 1.2.x jitterbuffer.

- Noah
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to