<div dir="ltr">Looks to that what you're dialing 404.... is not in your dialplan, you need to add an extesion for that, like:<div><br></div><div><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-family:'Times New Roman',serif"><span style="font-family:Consolas;color:rgb(31,73,125)"><font size="1"> <extension name="<a href="http://switch2voip.us/" target="_blank">o</a>utside_calls"><u></u><u></u></font></span></p><span class="im"><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-family:'Times New Roman',serif"><span style="font-family:Consolas;color:rgb(31,73,125)"><font size="1"> <condition field="destination_number" expression="^(404.*)$"></font></span></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-family:'Times New Roman',serif"><span style="font-family:Consolas;color:rgb(31,73,125)"><font size="1"> <action application="bridge" data="sofia/gateway/<a href="http://switch2voip.us/$1" target="_blank">switch2voip.us/$1</a>"/><u></u><u></u></font></span></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-family:'Times New Roman',serif"><span style="font-family:Consolas;color:rgb(31,73,125)"><font size="1"> </condition><u></u><u></u></font></span></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-family:'Times New Roman',serif"><span style="font-family:Consolas;color:rgb(31,73,125)"><font size="1"> </extension><u></u><u></u></font></span></p><div><span style="font-size:11pt;font-family:Consolas;color:rgb(31,73,125)"><br></span></div></span></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, Dec 28, 2014 at 11:42 PM, 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 lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">The full output from the “xml_locate dialplan” command is already in the previously pasted logfile.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Below is the dialplan that I created, in /usr/local/freeswitch/conf/dialplan/<a href="http://switch2voip.us" target="_blank">switch2voip.us</a>:<u></u><u></u></span></p><span class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"><?xml version="1.0" encoding="utf-8"?><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --><u></u><u></u></span></p></span><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <extension name="<a href="http://switch2voip.us" target="_blank">switch2voip.us</a>"><u></u><u></u></span></p><span class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <!-- If your provider does not provide ringback (180 or 183) you may simulate<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> ringback by uncommenting the following line. --><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <!-- action application="ringback" /--><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> <action application="bridge" data="sofia/gateway/<a href="http://switch2voip.us/$1" target="_blank">switch2voip.us/$1</a>"/><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> </condition><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:Consolas;color:#1f497d"> </extension><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p></span><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">My suspicion is that some other dialplan, other than my “<a href="http://switch2voip.us" target="_blank">switch2voip.us</a>” dialplan, is being invoked. My SIP Proxy is at 66.33.147.150. IP address “172.31.33.109” is the local/internal IP address for my AWS virtual cloud server. “4049392032” is a real phone number — not an extension.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Thanks,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">George<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><u></u> <u></u></span></p><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>David Villasmil Govea<br><b>Sent:</b> Sunday, December 28, 2014 5:05 PM</span></p><div><div class="h5"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable<u></u><u></u></div></div><p></p><div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">can you share your dialplan? It looks like you're dialing <u></u><u></u></p><div><p class="MsoNormal">"<span style="font-size:9.0pt;font-family:Consolas;color:#b1b100">To</span><span style="font-size:9.0pt;font-family:Consolas;color:black">: sip:</span><span style="font-size:9.0pt;font-family:Consolas;color:#cc66cc">4049392032</span><span style="font-size:9.0pt;font-family:Consolas;color:black">@<a href="http://172.31.33.109" target="_blank">172.31.33.109</a>"</span><u></u><u></u></p></div><div><p class="MsoNormal"><span style="font-size:9.0pt;font-family:Consolas;color:black">but have no extension for that...</span><u></u><u></u></p></div></div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">On Sun, Dec 28, 2014 at 10:55 PM, George F. Phelps <<a href="mailto:GeorgePhelps@gfphelps.com" target="_blank">GeorgePhelps@gfphelps.com</a>> wrote:<u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">New “pastebin” created:</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal" style="text-indent:.5in"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"><a href="http://pastebin.com/UwmgJGGg" target="_blank">http://pastebin.com/UwmgJGGg</a></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">George</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><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>David Villasmil Govea<br><b>Sent:</b> Sunday, December 28, 2014 4:04 PM</span><u></u><u></u></p><div><div><p class="MsoNormal"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable<u></u><u></u></p></div></div><div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal"><a href="http://pastebin.com/E4sqTLa4" target="_blank"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif"">http://pastebin.com/E4sqTLa4</span></a> doesn't show anything. Comes back with "<span style="font-size:10.0pt;font-family:"Segoe UI","sans-serif";color:#181818">This is a private paste. If you created this paste, please </span><a href="http://pastebin.com/login.php?ref=L0U0c3FUTGE0" target="_blank"><span style="font-size:10.0pt;font-family:"Segoe UI","sans-serif";color:#cc0033;text-decoration:none">login</span></a><span style="font-size:10.0pt;font-family:"Segoe UI","sans-serif";color:#181818"> to view it."</span><u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">On Sun, Dec 28, 2014 at 3:22 PM, George F. Phelps <<a href="mailto:GeorgePhelps@gfphelps.com" target="_blank">GeorgePhelps@gfphelps.com</a>> wrote:<u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Chris Tunbridge,</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">1) I made the updates to my configuration, as suggested in the “<a href="https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2</a>” link. I’m still not able to make a call to an outside number. A call to an extension connects, but there is still no audio.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">2) Extension x9161 is one of the default dialplan applications.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">3) Call failure log posted at: <a href="http://pastebin.com/E4sqTLa4" target="_blank">http://pastebin.com/E4sqTLa4</a></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Thanks,</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">George</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><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>Chris Tunbridge<br><b>Sent:</b> Saturday, December 27, 2014 2:30 AM</span><u></u><u></u></p><div><div><p class="MsoNormal"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable<u></u><u></u></p></div></div><div><div><p class="MsoNormal"> <u></u><u></u></p><div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">1) This is an issue with the NAT, likely on the freeswitch side, see instructions here: <a href="https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2" target="_blank">https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2</a> The important part is the external sip ip and external rtp ip. Without this calls will connect, but audio will not pass. I run dozens of servers on AWS without any issues as long as the external sip and rtp ip's are configured in the sip profile conf/sip_profiles/internal.xml<u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">2) Your issue you said was with extension <span style="font-family:"Verdana","sans-serif"">x9196, is this another sip endpoint or a dialplan application? If this is a sip endpoint, please make some adjustments to the conf/dialplan/default.xml to address extra extensions outside of the 10XX range.</span><u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-family:"Verdana","sans-serif"">3) Can you post a log here <a href="http://pastebin.freeswitch.org" target="_blank">http://pastebin.freeswitch.org</a> of a call attempt? My guess is that something's not matching the request, a complete log of a call attempt would help most here.</span><u></u><u></u></p></div><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">4) Glad to hear, its only used if you're using the JavaScript scripting engine for your scripts.</span><u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">On Fri, Dec 26, 2014 at 7:57 AM, George F. Phelps <<a href="mailto:GeorgePhelps@gfphelps.com" target="_blank">GeorgePhelps@gfphelps.com</a>> wrote:<u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Chris Tunbridge, et al.,</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">1) Freeswitch is running is running on an Amazon Web Services (AWS) Linux virtual cloud server. I am testing with Bria softphones (both Windows PC and Android smartphone) from my home network (behind a Netgear wireless router). The Freeswitch “show codecs” command indicates support for “codec, G.711 ulaw, CORE_PCM_MODULE” — which is the codec that I am using with Bria. I am able to successfully connect with Bria to my other VoIP services, such as VoIP.ms.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">2) I am using mostly a default configuration, i.e., extensions 1000 through 1019 are configured with updated passwords.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">3) This is my outbound dialplan. How do I know if this is the dialplan that is actually being used for dialing? It shows up in the “xml_locate dialplan” output — but as the very last entry. My guess is that Freeswitch is attempting to us some other (default, example?) gateway instead of my desired (<a href="http://switch2voip.us" target="_blank">switch2voip.us</a>) gateway.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <?xml version="1.0" encoding="utf-8"?></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <extension name="switch2voip"></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <!-- If your provider does not provide ringback (180 or 183) you may simulate</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> ringback by uncommenting the following line. --></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <!-- action application="ringback" /--></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> <action application="bridge" data="sofia/gateway/<a href="http://switch2voip.us/$1" target="_blank">switch2voip.us/$1</a>"/></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </condition></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </extension></span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">4) The “mod_v8” issue is now resolved. The module was not being built. I’m not sure why the downloaded default build/install files were not building it, but were attempting to load it. Sounds like a bug to me…</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">Thanks,</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d">George</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1f497d"> </span><u></u><u></u></p><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>Chris Tunbridge<br><b>Sent:</b> Thursday, December 25, 2014 9:25 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] SIP/2.0 480 Temporarily Unavailable</span><u></u><u></u></p><div><div><p class="MsoNormal"> <u></u><u></u></p><div><div><div><div><p class="MsoNormal">1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration<u></u><u></u></p></div><p class="MsoNormal">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)<u></u><u></u></p></div><p class="MsoNormal">3) Do you have an outbound route configured that matches your dial string?<u></u><u></u></p></div><p class="MsoNormal">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<u></u><u></u></p></div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">On Wed, Dec 24, 2014 at 9:39 AM, George F. Phelps <<a href="mailto:GeorgePhelps@gfphelps.com" target="_blank">GeorgePhelps@gfphelps.com</a>> wrote:<u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I am debugging a new/initial Freeswitch configuration.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I believe that I have successfully registered with my VoIP provider — “State=REGED”.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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?</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">I am getting this critical error on startup. Problem #4.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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**</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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?</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">Version info:</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></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)</span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif""> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">Thanks!</span><u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p></div></div><p class="MsoNormal"><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a 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