<div dir="ltr"><div><div><div>1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration<br></div>2) If you're using default configs, its configured to look for extensions 10XX, you can see this in conf/dialplan/default.xml (and in conf/dialplan/public.xml for calls coming from the outside)<br></div>3) Do you have an outbound route configured that matches your dial string?<br></div>4) This just means the module wasn't configured, you can comment out the line in conf/autoload_configs/modules.conf.xml find the line that says mod_v8 and put a <!-- at the beginng and a -> at the end<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <span dir="ltr"><<a href="mailto:GeorgePhelps@gfphelps.com" target="_blank">GeorgePhelps@gfphelps.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I am debugging a new/initial Freeswitch configuration.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I believe that I have successfully registered with my VoIP provider — “State=REGED”.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I am able to dial from one extension (x1000) to a different extension (x1001), but after answering, there is NO AUDIO at either end of the call. Problem #1.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">When I test call to extension x9196, for example, I get an immediate hang-up and SIP response of “SIP/2.0 480 Temporarily Unavailable”. Problem #2. Do I have to do anything to enable calling to x9196?<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">And when I attempt to call an external phone number via my VoIP provider, I get the same immediate hang-up and SIP response. Problem #3.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I am getting this critical error on startup. Problem #4.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal" style="text-indent:.5in"><span style="font-family:"Verdana","sans-serif"">2014-12-24 11:29:09.869357 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_v8.so<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal" style="text-indent:.5in"><span style="font-family:"Verdana","sans-serif"">**/usr/local/freeswitch/mod/mod_v8.so: cannot open shared object file: No such file or directory**<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">Any suggestions as to what configuration might be wrong? Or how I can get additional debug information?<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">Version info:<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal" style="text-indent:.5in"><span style="font-family:"Verdana","sans-serif"">FreeSWITCH Version 1.5.15b+git~20141222T221908Z~067cb0f0f2~64bit (git 067cb0f 2014-12-22 22:19:08Z 64bit)<u></u><u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">Thanks!<u></u><u></u></span></p><p class="MsoNormal"><u></u> <u></u></p></div></div><br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div>