<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:x="urn:schemas-microsoft-com:office:excel" xmlns:p="urn:schemas-microsoft-com:office:powerpoint" xmlns:oa="urn:schemas-microsoft-com:office:activation" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";
color:black;}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
pre
{mso-style-priority:99;
mso-style-link:"HTML Preformatted Char";
margin:0in;
margin-bottom:.0001pt;
font-size:10.0pt;
font-family:"Courier New";
color:black;}
span.HTMLPreformattedChar
{mso-style-name:"HTML Preformatted Char";
mso-style-priority:99;
mso-style-link:"HTML Preformatted";
font-family:Consolas;
color:black;}
span.EmailStyle19
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page Section1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.Section1
{page:Section1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body bgcolor=white lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I had a lot of problems with garbled IAX calls (even when
calling into just the IVR). The problem was compacted when I would bridge an
incoming IAX call to an outgoing SIP call, though that may be a fault of
Asterisk. Since using SIP everything has been working perfectly. I never had
any real problems with dropping calls (that weren’t on my end). However, I
don’t use IAX anymore, so I am not aware of any current issues.<o:p></o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Christopher
Robinson<br>
<b>Sent:</b> Wednesday, August 08, 2007 4:54 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] VoicePulse Connect<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Wes, I'm working through some issues with IAX and Voicepulse
right now. It was regarding dropped inbound calls. I was able to
put my server into a different location though, and so far the issues have
disappeared so hopefully it was a network problem somewhere between us.
Just curious what problems you encountered as I would prefer to use IAX if
possible.<br>
<br>
John, I've tried a few services, and Voicepulse was the clear winner for
me. I still have two other services in my dialplan for failover, but
Voicepulse will remain the primary for now. The voice quality has been
very good, and their technical support has been absolutely fantastic for a
no-charge service.<br>
<br>
Wes Baehr wrote: <o:p></o:p></p>
<pre>If you cannot afford any dropped calls or poor audio quality, you need a PRI<o:p></o:p></pre><pre>or POTS connection. It doesn't matter how great the carrier is, the Internet<o:p></o:p></pre><pre>is an unreliable medium.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>2-3 times VoicePulse has had issues with incomings calls ringing busy. Once<o:p></o:p></pre><pre>incoming calls were all garbled on my end, although the customer could hear<o:p></o:p></pre><pre>me fine.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Generally, the outbound service is reliable. However, you should have a<o:p></o:p></pre><pre>backup carrier anyway.<o:p></o:p></pre><pre> <o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>-----Original Message-----<o:p></o:p></pre><pre>From: <a
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><o:p></o:p></pre><pre>[<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of John Meksavan<o:p></o:p></pre><pre>Sent: Wednesday, August 08, 2007 1:09 PM<o:p></o:p></pre><pre>To: <a
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><o:p></o:p></pre><pre>Subject: Re: [asterisk-users] VoicePulse Connect<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Wes,<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> What kind of service outages did you experienced?<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> This would use for my office and I cannot afford for any dropped calls or<o:p></o:p></pre><pre>poor audio quality, when talking to customers.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>-John<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> <o:p></o:p></pre>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><pre>From: "Wes Baehr" <a
href="mailto:wbaehr@abilitybusiness.com"><wbaehr@abilitybusiness.com></a><o:p></o:p></pre><pre>Reply-To: Asterisk Users Mailing List - Non-Commercial <o:p></o:p></pre><pre>Discussion<a
href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a><o:p></o:p></pre><pre>To: "'Asterisk Users Mailing List - Non-Commercial <o:p></o:p></pre><pre>Discussion'"<a
href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a><o:p></o:p></pre><pre>Subject: Re: [asterisk-users] VoicePulse Connect<o:p></o:p></pre><pre>Date: Wed, 8 Aug 2007 12:55:29 -0400<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>John,<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Voicepulse Connect has been great to me. I've been using it for over a <o:p></o:p></pre><pre>year now, and do not have any major complaints, except that there are <o:p></o:p></pre><pre>no printable receipts for credit card transactions. SIP is also the <o:p></o:p></pre><pre>preferable protocol, as IAX seems to have some issues. Customer service <o:p></o:p></pre><pre>is usually pretty good, and there have been very few (although a <o:p></o:p></pre><pre>couple) problems with service outages.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>-----Original Message-----<o:p></o:p></pre><pre>From: <a
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><o:p></o:p></pre><pre>[<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of John <o:p></o:p></pre><pre>Meksavan<o:p></o:p></pre><pre>Sent: Wednesday, August 08, 2007 12:30 PM<o:p></o:p></pre><pre>To: <a
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><o:p></o:p></pre><pre>Subject: [asterisk-users] VoicePulse Connect<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Asterisk Users,<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> Has anybody use Voicepulse Connect for Asterisk?<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> I am trying to cover all my bases because in the past, I got burned <o:p></o:p></pre><pre>with poor quality of service, along with failed DTMF tones with 3 <o:p></o:p></pre><pre>different SIP Providers (Vitelity, Broadvoice, and Teliax).<o:p></o:p></pre><pre><o:p> </o:p></pre><pre> I am running Asterisk 1.2.13 on the Debian Etch system, using the <o:p></o:p></pre><pre>SIP protocol. Any insights would be great. Thanks.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>-John<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>_________________________________________________________________<o:p></o:p></pre><pre>Tease your brain--play Clink! Win cool prizes!<o:p></o:p></pre><pre><a
href="http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2">http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre> <o:p></o:p></pre></blockquote>
<pre><o:p> </o:p></pre><pre>_________________________________________________________________<o:p></o:p></pre><pre>Find a local pizza place, movie theater, and more.then map the best route! <o:p></o:p></pre><pre><a
href="http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theater">http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theater</a><o:p></o:p></pre><pre>&cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&scene=95060<o:p></o:p></pre><pre>7&encType=1&FORM=MGAC01<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre> <o:p></o:p></pre>
<p class=MsoNormal><o:p> </o:p></p>
</div>
</body>
</html>