Hi Veselin,<br><br>You can verify SDP and RTP by running protocol analyzer such Ethereal, if you need instruction you can follow this link. <a href="http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html">http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html
</a><br><br>While TDM side, I think he is referring to the card, if the card is faulty or not.<br><br>Best Regards,<br>Joanna<br><br><div class="gmail_quote">On Dec 1, 2007 9:27 AM, Veselin Kantsev <<a href="mailto:veselin@campbell-lange.net">
veselin@campbell-lange.net</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Thank you much for the prompt reply Salvatore.<br>
<br>Would you have the time to explain further how should I go for verifying<br>that SDP and RTP are OK.<br>Also what is reffered to as the TDM site.<br><font color="#888888"><br>Veselin<br></font><div class="Ih2E3d"><br>
On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:<br>> Take a packet capture of your VoIP segment and verify that the SDP is<br>> correct and that the RTP is making it to the correct places. If all that
<br>> looks good and this is a straight out quality problem, then you need to<br>> figure out if it's happening on the voip side or on the TDM side. You should<br>> make calls (with captures) VoIP to Voip passing the media through your
<br>> asterisk and also try routing a tdm call in and back out. If you have the<br>> equipment, take a mos score of the TDM loop.<br>><br>> Without any of the above, you will not be able to isolate the issue.<br>
><br>> --------------------------------------------------<br>> Salvatore Giudice<br>> <a href="mailto:Salvatore.Giudice@VoIPSecurityTraining.com">Salvatore.Giudice@VoIPSecurityTraining.com</a><br>><br>> VoIP Security Training, LLC
<br>> <a href="http://VoIPSecurityTraining.com" target="_blank">http://VoIPSecurityTraining.com</a><br>><br>> 848 N. Rainbow Blvd. #1676<br>> Las Vegas, NV 89107<br>> Phone: (617) 959-7625<br>> Fax: (214) 279-2906
<br>><br>><br>> -----Original Message-----<br>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">
asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Veselin<br>> Kantsev<br>> Sent: Friday, November 30, 2007 2:47 PM<br>> To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com
</a><br>> Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality<br>><br></div><div><div></div><div class="Wj3C7c">> Hello,<br>> I have an Asterisk running with a Sangoma A200 card with Hardware Echo
<br>> cancelling connected to the UK PSTN.<br>> If a PSTN call comes in, voice both ways is OK, however if an outgoing<br>> call over the PSTN is made I can hear the other party OK but they can<br>> not, they can barely understand what I am saying, my voice is unclear
<br>> fading and skipping.<br>> Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2<br>> are OK too. I've tried gsm/ulaw/alaw codecs so far.<br>> Tried disabling the echo cancelling as well.
<br>><br>> Any suggestions will be greatly appreciated.<br>><br>><br>> Regards,<br>> Veselin<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by
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