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 <[EMAIL PROTECTED]> 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 > > [EMAIL PROTECTED] > > > > 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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Veselin > > Kantsev > > Sent: Friday, November 30, 2007 2:47 PM > > To: [email protected] > > 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 > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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