Haudy Kazemi wrote: > On Aug 2 2007, John Meksavan wrote: > > >> Asterisk Users, >> >> I recently ran into some problems with the quality of service with >> Teliax. >> This occurred on August 1, 2007 with a dropped outbound call, audio >> quality isse on the callee side- not hearing me well on callee side, and >> sending DTMF tones (configured for RFC2833). Am I the only Teliax >> customer having this problem? >> >> It seems like when I am ready to go live with my Asterisk PBX System, I >> run into quality of service issues with the SIP provider. Who should I go >> with that would guarantee me quality service just like an analog line? >> > > VoIP is susceptible to packet delivery problems anywhere between your PBX > and your SIP provider's PRI lines/termination point. If you have direct SIP > PBX to SIP PBX calls, then your problems can be anywhere on the Internet > path between the sites. The only workaround that I know of is having your > ISP be your SIP provider, so that your SIP packets only cross your ISP's > own network to its termination point, and do not cross the public Internet. > This way QoS can work from your office to your ISP's office to make sure > you maintain reliability. > > I have not personally used iTEL-ip's 'iTEL Voice Service', but others have > said, as do their own notes that their network QoS is effective at > maintaining call quality. When I contacted them, their pricing for a 'QoS > private IP backbone for voice and data' was $618/month for a full 1.5mbps > T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending > on contract length. Per minute rates were $.03. > > When I ran the numbers, it appeared that a regular full T1 + a regular full > PRI would be only slightly more. A major tradeoff comes in the physical > location flexibility you get with SIP over traditional phone lines in the > case you need to move an office (although physically moving the phones to a > non iTEL-ip data line would mean you're not getting their Qos). > > iTEL-ip's 'iTEL Voice Service' > http://www.itelconnect.com/default.aspx?type=t§ion=iTEL-ipVoiceService&selection=16 > > http://wiki.pbxnsip.com/index.php/ITEL-ip > > -hk > > _______________________________________________ > --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 > On the original problem of missed DTMF set dtmfmode=info in your sip.conf.
Anthony _______________________________________________ --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