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: asterisk-users@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 > > _______________________________________________ > --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