If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium.
2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John >From: "Wes Baehr" <[EMAIL PROTECTED]> >Reply-To: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users@lists.digium.com> >To: "'Asterisk Users Mailing List - Non-Commercial >Discussion'"<asterisk-users@lists.digium.com> >Subject: Re: [asterisk-users] VoicePulse Connect >Date: Wed, 8 Aug 2007 12:55:29 -0400 > >John, > >Voicepulse Connect has been great to me. I've been using it for over a >year now, and do not have any major complaints, except that there are >no printable receipts for credit card transactions. SIP is also the >preferable protocol, as IAX seems to have some issues. Customer service >is usually pretty good, and there have been very few (although a >couple) problems with service outages. > > > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of John >Meksavan >Sent: Wednesday, August 08, 2007 12:30 PM >To: asterisk-users@lists.digium.com >Subject: [asterisk-users] VoicePulse Connect > >Asterisk Users, > > Has anybody use Voicepulse Connect for Asterisk? > > I am trying to cover all my bases because in the past, I got burned >with poor quality of service, along with failed DTMF tones with 3 >different SIP Providers (Vitelity, Broadvoice, and Teliax). > > I am running Asterisk 1.2.13 on the Debian Etch system, using the >SIP protocol. Any insights would be great. Thanks. > > >-John > >_________________________________________________________________ >Tease your brain--play Clink! Win cool prizes! >http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 > > >_______________________________________________ >--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 _________________________________________________________________ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theater &cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&scene=95060 7&encType=1&FORM=MGAC01 _______________________________________________ --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