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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Chris Tunbridge, et al.,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></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.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></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.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></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 (switch2voip.us) gateway.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <?xml version="1.0" encoding="utf-8"?><o:p></o:p></span></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 --><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <extension name="switch2voip"><o:p></o:p></span></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})$"><o:p></o:p></span></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<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> ringback by uncommenting the following line. --><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <!-- action application="ringback" /--><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <action application="bridge" data="sofia/gateway/switch2voip.us/$1"/><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> </condition><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> </extension><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></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…<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Thanks,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>George<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></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"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <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<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><div><div><p class=MsoNormal>1) Sounds like NAT issue possibly, or incorrect codecs, please elaborate on your topology and configuration<o:p></o:p></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)<o:p></o:p></p></div><p class=MsoNormal>3) Do you have an outbound route configured that matches your dial string?<o:p></o:p></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<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></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:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'>I am debugging a new/initial Freeswitch configuration.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'>I believe that I have successfully registered with my VoIP provider — “State=REGED”.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'>I am getting this critical error on startup. Problem #4.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'>Version info:</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;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><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-family:"Verdana","sans-serif"'>Thanks!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div><p class=MsoNormal><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><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>