<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>This is something you can do in standard dialplan xml. See continue_on_fail. Also check out mod_lcr for doing least cost routing and backing it up with another option if one fails.<br><br>Sent from my iPhone</div><div><br>On Jan 13, 2013, at 10:22 PM, "Sean Devoy" <<a href="mailto:sdevoy@bizfocused.com">sdevoy@bizfocused.com</a>> wrote:<br><br></div><blockquote type="cite"><div><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"><meta name="Generator" content="Microsoft Word 14 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri","sans-serif";}
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;}
span.EmailStyle17
{mso-style-type:personal-compose;
font-family:"Calibri","sans-serif";
color:windowtext;}
.MsoChpDefault
{mso-style-type:export-only;
font-family:"Calibri","sans-serif";}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></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]--><div class="WordSection1"><p class="MsoNormal">Hi,<o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">This question stems from the fact that I cannot find a VOIP Service Provider (VSP) that can connect outgoing calls truly reliably. I have customers who expect calls to go through just like good old ma-bell did –100% of the time barring natural disasters. You have to admit those guys set the bar pretty high for us.<o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">I was using Voip Innovations with decent success, but my users complain every 2 weeks or so and I send the trace to VI. The standard response is always “We have updated our route tables for that area.” I switched to Vitelity where I was assured they have redundant paths and this could never happen! I have only confirmed it happening twice in the last 3 weeks! Of course one of those was during the demo of the new service I had just installed for a new customer. <o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">So, I give up. My new approach is “VSPs cannot deliver consistent highly reliable service – work around it!”<o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">I know I want a dial plan that will check the bridge status and redial using a different gateway (VSP) if it failed (not busy or no answer – failure). I am pretty sure the best/easy approach is a JavaScript “dialer” script to handle this.<o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">I have looked in the wiki for such a sample. If it exists I have missed it. So rather than “re-inventing the wheel” I am reaching out to you all. Does anyone have a dialing script that checks for failure and redials on failure or just continues to voicemail on no answer? I will happily test and tweak it and add it to the wiki if you have something to get me started!!! If not I shall take the plunge and get started! My ideal solution will dial, play a message and redial on failure, maybe 2 or three times or play a failure message and email me on excess errors. I am leaning toward “Call congestion, rerouting. Please stand by” while the new bridge is being attempted.<o:p></o:p></p><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal">Thanks,<o:p></o:p></p><p class="MsoNormal">Sean<o:p></o:p></p></div></div></blockquote><blockquote type="cite"><div><span>_________________________________________________________________________</span><br><span>Professional FreeSWITCH Consulting Services:</span><br><span><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a></span><br><span><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a></span><br><span></span><br><span>FreeSWITCH-powered IP PBX: The CudaTel Communication Server</span><br><span><a href="http://www.cudatel.com">http://www.cudatel.com</a></span><br><span></span><br><span>Official FreeSWITCH Sites</span><br><span><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></span><br><span><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a></span><br><span><a href="http://www.cluecon.com">http://www.cluecon.com</a></span><br><span></span><br><span>FreeSWITCH-users mailing list</span><br><span><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a></span><br><span><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></span><br><span>UNSUBSCRIBE:http://<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">lists.freeswitch.org/mailman/options/freeswitch-users</a></span><br><span><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></span><br></div></blockquote></body></html>