[Asterisk-Users] need help troubleshooting clipping and garbled VOIP calls

Thomas Kenyon digium at sanguinarius.co.uk
Fri Jun 30 01:56:15 MST 2006


T. Shaw wrote:

> > Hello all,
> > I have a problem with call quality with my Asterisk setup. I'm doing
> > VOIP only so far, but have a zaptel TDM400P in the box not being used.
> > The problem i'm having is that when calls are placed, connected, and
> > the far-end is reporting that they are experiencing clipping, choppy,
> > and garbled voice conversations. So bad that we have to resort to
> > using our cell phones. This entire setup is still being built, but any
> > phone attached is experiencing this. Call volume is almost nil (under
> > 20 total incoming calls a day). This is a small business setup. The
> > server is used exclusively for Asterisk, so it isn't a fileserver, or
> > anything else.
> >
> > The setup is as such:
> >
> > ipphone  <--->cisco 2900XL switch <----> Cisco 2621 router <---> dsl
> > modem <-->DSL <---> VOIPprovider
> >
> > I've configured the switch and the router to set priority and qos to
> > prioritize voice traffic above data.
> > Funny thing is, there is not data REALLY hitting the network. I have
> > setup 2 vlans, data vlan, and voice vlan. There are two work stations
> > on the network, and neither is being used to hit the internet heavily
> > (office is still being setup).
> >
> > Any pointers or suggestions anyone have for me as to were to look for
> > this poor quality?
> > It seems only the Far-end (called party), is hearing this and not the
> > calling party.
> >
> > I haven't tried switching out the phones because we only have 1 type,
> > and any of the phones i used exhibit these problems. I will try
> > softphones to see if it is truly a "networking" issue or Phone issue.
> >
> > Is anyone using a cisco 2900 switch or router and care to provide
> > config samples of their COS/QOS setup?
> >
> > Thanks!
> >
> > Terrelle Shaw
> >
>   

I've got a similar setup (which does have a TDM card and voip incoming
and outgoing), for some reason an IAX provider (which provides most of
our calls incoming and outgoing) has this problem, whereas a different
SIP one doesn't seem to.

I have checked my traffic shaping script, and everything seems fine, the
same provider works flawlesly from home, with a simliar setup (only
without a timing source and a cable modem).

I'd be very interested to see what you find out.


>




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