[asterisk-users] Random 'no audio' problem

[Matt]

I've figured out the problem, alaredy, and posted in another thread,
but no one seems to have an answer yet.

It is a problem in the IAX trunk.  If I turn the jitterbuffer on, I
get one-way-audio when I put someone on hold.  If I turn the
jitterbuffer off... I still have two way audio.  THis is running 1.2.x

Anyone have any bright ideas?

On 11/12/06, Jordi Nelissen <jordi.nelissen at escaux.com> wrote:
> Matt,
>
> as a start, what I can advise you is to take a tethereal trace and try
> to reproduce the problem.
>
> "nohup tethereal host a.b.c.d -s2000 -w /tmp/yourtrace.cap &"
>
> Where a.b.c.d is the IP address of your IP phone. You can then analyse
> the trace and at least see if the asterisk box is sending AND receiving
> RTP traffic to and from the phone.
>
> We have seen some issues in the past with 'no audio' or 'unidirectional
> audio' due to wrong firmware versions in SIP phones or due to ethernet
> switch instability, even on a cisco 3560 switches.
>
> Hope this helps,
>
> Jordi
>
> Matt wrote:
> > I have no idea.. that sounds like your Internet connection is going
> > down and leaving you for a bit and then coming back.   My issue is a
> > local network connection, no public Internet... or you can even call
> > in from outside on the PSTN and the audio, both ways, will just stop.
> >
> > On 11/3/06, Zeeshan Zakaria <zishanov at gmail.com> wrote:
> >> I have the same problem with IAX trunk and SIP extensions. Now I think
> >> its
> >> the IAX. I never had this problem om SIP trunk. Am I right?
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> --
> Jordi Nelissen
>
> E S C A U X
> Business IP Telephony
>
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