It should be another simple issue. Check your default.xml dialplan. It has entries for 1000-1020 numbers, but you must similarly add for your new #s.<div><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
2012/11/2 Rob Boutin <span dir="ltr"><<a href="mailto:rjb@contactbeep.com" target="_blank">rjb@contactbeep.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Alright so my phones have ring tones and can place calls to the predefined extensions (1000 – 1020) but upon trying to call one another it always says “temporarily unavailable” there is no difference in the actual extension files between the working ones and the new ones I defined but the server doesn’t seem able to connect the calls from my new extensions to my other new extensions for some reason. <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Any ideas on this would be very helpful, and I feel like whatever this is should be the last issue I face.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks again,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">RJB <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></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"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Rob Boutin<br>
<b>Sent:</b> Friday, November 02, 2012 8:54 AM</span></p><div><div class="h5"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Just starting<u></u><u></u></div></div><p></p></div></div><div><div class="h5">
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thank you, <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Using this I was able to realize the embarrassingly simple solution.<u></u><u></u></span></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">My XML editor at some point during my setup lost permission to change files in the FreeSWITCH folder and wasn’t throwing errors for some reason. <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks again,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">RJB<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><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"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Vitalie Colosov<br>
<b>Sent:</b> Thursday, November 01, 2012 3:44 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Just starting<u></u><u></u></span></p></div><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">
If you could shutdown all devices and then capturing SIP Trace while bringing up the device with #300 (or any # except #301) using Wireshark, it will definitely add more info for investigation.<u></u><u></u></p><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><div><p class="MsoNormal">2012/11/1 Rob Boutin <<a href="mailto:rjb@contactbeep.com" target="_blank">rjb@contactbeep.com</a>><u></u><u></u></p><div><div><p class="MsoNormal">Hello,<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">I am just starting out with FreeSWITCH and I am researching it for a project at work that will hopefully end up with full implementation of FreeSWITCH and had a few short questions as I am having difficulties getting it off the ground.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system.<u></u><u></u></p><p class="MsoNormal">My tiny test network of phones includes an xlite softphone on the windows machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone. They are being routed with a simple Linksys Etherfast router.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">My issue is that only one phone at a time will ever be registered and able to make calls. I have four extensions configured with identical settings 300 – 303. The peculiar thing is only the phone assigned to 301 will ever have a dial tone and be able to place calls. It doesn’t matter which device I move 301 to it is only 301 that ever works. I was hoping to get some light shed on this, as it is likely related to some simple setting I over looked and would like to continue with my research.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Thanks in advance,<u></u><u></u></p><p class="MsoNormal">RJB<u></u><u></u></p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_________________________________________________________________________<br>
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<br></blockquote></div><br></div>