[asterisk-users] Outgoing PSTN calls , unusable voice quality

[Joanna Liza Mariazeta]

Hi Veselin,

You can verify SDP and RTP by running protocol analyzer such Ethereal, if
you need instruction you can follow this link.
http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html

While TDM side, I think he is referring to the card, if the card is faulty
or not.

Best Regards,
Joanna

On Dec 1, 2007 9:27 AM, Veselin Kantsev <veselin at campbell-lange.net> wrote:

> Thank you much for the prompt reply Salvatore.
>
> Would you have the time to explain further how should I go for verifying
> that SDP and RTP are OK.
> Also what is reffered to as the TDM site.
>
> Veselin
>
> On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:
> > Take a packet capture of your VoIP segment and verify that the SDP is
> > correct and that the RTP is making it to the correct places. If all that
> > looks good and this is a straight out quality problem, then you need to
> > figure out if it's happening on the voip side or on the TDM side. You
> should
> > make calls (with captures) VoIP to Voip passing the media through your
> > asterisk and also try routing a tdm call in and back out. If you have
> the
> > equipment, take a mos score of the TDM loop.
> >
> > Without any of the above, you will not be able to isolate the issue.
> >
> > --------------------------------------------------
> > Salvatore Giudice
> > Salvatore.Giudice at VoIPSecurityTraining.com
> >
> > VoIP Security Training, LLC
> > http://VoIPSecurityTraining.com
> >
> > 848 N. Rainbow Blvd. #1676
> > Las Vegas, NV 89107
> > Phone: (617) 959-7625
> > Fax: (214) 279-2906
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
> > Kantsev
> > Sent: Friday, November 30, 2007 2:47 PM
> > To: asterisk-users at lists.digium.com
> > Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality
> >
> > Hello,
> > I have an Asterisk running with a Sangoma A200 card with Hardware Echo
> > cancelling connected to the UK PSTN.
> > If a PSTN call comes in, voice both ways is OK, however if an outgoing
> > call over the PSTN is made I can hear the other party OK but they can
> > not, they can barely understand what I am saying, my voice is unclear
> > fading and skipping.
> > Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2
> > are OK too. I've tried gsm/ulaw/alaw codecs so far.
> > Tried disabling the echo cancelling as well.
> >
> > Any suggestions will be greatly appreciated.
> >
> >
> > Regards,
> > Veselin
> >
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