<div dir="ltr"><div class="gmail_default" style="color:#660000"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 29, 2015 at 11:02 PM, Murthy Gandikota <span dir="ltr"><<a href="mailto:murthy64@hotmail.com" target="_blank">murthy64@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div><div dir="ltr"><br><br><div><hr>Date: Wed, 29 Jul 2015 11:47:19 -0500<br>From: <a href="mailto:sgriepentrog@digium.com" target="_blank">sgriepentrog@digium.com</a><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><div><div class="h5"><br>Subject: Re: [asterisk-users] Windows Asterisk Help<br><br><div dir="ltr"><div><br><div>On Wed, Jul 29, 2015 at 10:16 AM, John Novack <span dir="ltr"><<a href="mailto:jnovack@stromberg-carlson.org" target="_blank">jnovack@stromberg-carlson.org</a>></span> wrote:<br><blockquote style="border-left:1px #ccc solid;padding-left:1ex">
  
    
  
  <div><div><div>
    <br>
    <br>
    <div>Murthy Gandikota wrote:<br>
    </div>
    <blockquote>
      
      <div dir="ltr"><br>
        <br>
        <div>
          <hr>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
          From: <a href="mailto:webaccounts173@jgoettgens.de" target="_blank">webaccounts173@jgoettgens.de</a><br>
          Date: Wed, 29 Jul 2015 16:11:31 +0200<br>
          Subject: Re: [asterisk-users] Windows Asterisk Help<br>
          <br>
          <div><br>
          </div>
          <blockquote>
            
            <div dir="ltr"><br>
              <div>Downloaded latest version of Asterisk from <a href="http://www.asteriskwin32.com" target="_blank">www.asteriskwin32.com</a>
                and installed on Windows 7.</div>
              <div><br>
              </div>
              <div>Here  is my sip.conf</div>
              <div><br>
              </div>
              <div>
                <div>[general]</div>
                <div>context = demo  ;              Default context for
                  incoming calls</div>
                <div>bindport = 5060  ;              UDP Port to bind to
                  (SIP standard port is 5060)</div>
                <div>bindaddr = 0.0.0.0  ;              IP address to
                  bind to (0.0.0.0 binds to all)</div>
                <div>srvlookup = yes  ;              Enable DNS SRV
                  lookups on outbound calls</div>
                <div>context=incoming </div>
                <div>disallow=all </div>
                <div>allow=ulaw </div>
                <div>allow=alaw </div>
                <div>allow=g729 </div>
                <div>allow=g723 </div>
                <div>externip=72.220.28.226 </div>
                <div>localnet=192.168.0.0 </div>
                <div>nat=yes </div>
                <div>maxexpiry=15</div>
                <div>minexpiry=14</div>
                <div>;rtautoclear=no</div>
                <div>;autofallthrough=yes</div>
                <div><br>
                </div>
                <div>register
                  =>16194077214:<<password>@<a href="http://69.59.234.67:5060/202" target="_blank">69.59.234.67:5060/202</a></div>
                <div><br>
                </div>
                <div>[authentication]</div>
                <div>[3000]</div>
                <div>type = friend</div>
                <div>context = default</div>
                <div>username = 3000</div>
                <div>host = dynamic</div>
                <div>mailbox = 3000</div>
                <div>dtmfmode = rfc2833</div>
                <div>[3001]</div>
                <div>type = friend</div>
                <div>context = default</div>
                <div>username = 3001</div>
                <div>host = dynamic</div>
                <div>mailbox = 3001</div>
                <div>dtmfmode = rfc2833</div>
                <div><br>
                </div>
                <div>[3002]</div>
                <div>type = friend</div>
                <div>username = 3002</div>
                <div>context = default</div>
                <div>host = dynamic</div>
                <div>mailbox = 3002</div>
                <div>dtmfmode = rfc2833</div>
                <div><br>
                </div>
                <div>[vonage-out]</div>
                <div><br>
                </div>
                <div>username=16194077214</div>
                <div><br>
                </div>
                <div>type=friend</div>
                <div><br>
                </div>
                <div>secret=<<password>></div>
                <div><br>
                </div>
                <div>port=5061</div>
                <div><br>
                </div>
                <div>nat=yes</div>
                <div><br>
                </div>
                <div>host=69.59.234.67</div>
                <div><br>
                </div>
                <div>fromuser=16194077214</div>
                <div><br>
                </div>
                <div>fromdomain=69.59.234.67</div>
                <div><br>
                </div>
                <div>dtmfmode=rfc2833</div>
                <div><br>
                </div>
                <div>auth=md5</div>
                <div><br>
                </div>
                <div>[vonage202]</div>
                <div><br>
                </div>
                <div>username=16194077214</div>
                <div><br>
                </div>
                <div>;type=friend</div>
                <div>type=peer</div>
                <div>;type=user</div>
                <div><br>
                </div>
                <div>secret=<<password>></div>
                <div><br>
                </div>
                <div>port=5061</div>
                <div><br>
                </div>
                <div>nat=yes</div>
                <div><br>
                </div>
                <div>insecure=port,invite</div>
                <div><br>
                </div>
                <div>host=69.59.234.67</div>
                <div><br>
                </div>
                <div>fromuser=16194077214</div>
                <div><br>
                </div>
                <div>fromdomain=69.59.234.67</div>
                <div><br>
                </div>
                <div>;dtmfmode=inband</div>
                <div><br>
                </div>
                <div>context=from-pstn</div>
                <div><br>
                </div>
                <div>canreinvite=no</div>
                <div><br>
                </div>
                <div>;auth=md5</div>
                <div>disallow=all</div>
                <div>allow=ulaw </div>
                <div>;allow=alaw </div>
                <div>;allow=g729 </div>
                <div>;allow=g723 </div>
              </div>
              <div><br>
              </div>
              <div>Here is my extensions.conf</div>
              <div><br>
              </div>
              <div>
                <div>[from-pstn]</div>
                <div>;exten => 16194077214,1,verbose(0, hello)</div>
                <div>exten => 16194077214,1,Answer;</div>
                <div>exten => 16194077214,n,SayUnixTime()</div>
                <div>exten => 16194077214,n,Hangup</div>
                <div><br>
                </div>
                <div><br>
                </div>
              </div>
              <div>I am able to connect with Asterisk on the first try
                after fresh load, but not on the subsequent tries.</div>
              <div>I have to re-reload sip.conf and extensions.conf to
                connect with Asterisk. Looking at the logs, it seems
                like a registration issue.  So I set minexpirty and
                maxexpirty that seems to have no effect.  can post the
                logs, if someone wants me to.</div>
              <div><br>
              </div>
              <div>Your kind help is appreciated.</div>
              <div><br>
              </div>
              <div>Best regards</div>
              <div>murthy</div>
              <div><br>
              </div>
              <div><br>
              </div>
            </div>
            <br>
            <fieldset></fieldset>
            <br>
          </blockquote>
          <a href="http://www.asteriskwin32.com" target="_blank">www.asteriskwin32.com</a>
          hosts only a very very old version of Asterisk
          (1.2.something). What speaks against setting up a small
          virtual machine to host a recent version of Asterisk?<br>
          <br>
          jg<br>
          <br>
          You have a point. My SIP provider at the moment is Vonage
          which I can't access from work (some security issue:)</div>
        <div>So I am confined to testing from home and I don't have any
          other machine to spare. If there is no other way</div>
        <div>to trouble-shoot the problem, I will have to do what you
          suggest. </div>
        <div><br>
        </div>
        <div>Thanks & Regards</div>
        <div>murthy</div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
    </blockquote></div></div>
    For very little $$$ you could obtain an HP thin client, load a
    modern version of Asterisk using AstLinux, and leave your Win 7
    machine to do what it does best ( which is certainly NOT Asterisk )<br>
    Once installed, it can be completely controlled and configured
    remotely over your home LAN, consumes very little power, has a
    universal power supply, consumes little power and no noisy fans.<br>
    HP5720 units can be had off eBay for $20-30 US. Even with shipping
    to your country, really low cost solution much more in the
    mainstream.<br>
    AstLinux uses standard Asterisk confs. The GUI is used for
    management and editing, and doesn't  use the difficult to
    troubleshoot  and quirky overlays of a TrixBox or FreePBX<br>
    Check out the astlinux website for more details<span><font color="#888888"><br>
    <br>
    John Novack<br>
    <br>
    <pre>-- 

Dog is my Co-pilot</pre>
  </font></span></div>

<br>--<br>
_____________________________________________________________________<br>
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<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><div><div style="color:rgb(102,0,0)">​Another option (assuming your computer has enough ram and disk space) is to run a copy of Linux in Vmware Player (which is available for free).  It allows you to run the Linux environment in a virtual computer as if it was an application on windows.  Then you can test the most recent release of Asterisk (version 13 at the moment).​</div><br></div>-- <br><div><div dir="ltr"><img alt="Digium logo" src="https://my.digium.com/images/graphics/digium_RGB_signature.gif" width="288" height="50" style="color:rgb(0,0,0);font-family:Arial,Helvetica,sans-serif;font-size:12px"><div>Scott Griepentrog<br>Digium, Inc · Software Developer<br>445 Jan Davis Drive NW · Huntsville, AL 35806 · US<br>direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090<br>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> · <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br></div></div></div>
</div></div>
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></div></div><div><br></div><div><br></div><div><br></div><div><br></div><div>Many thanks for your kind replies. Here is what I have done:</div><div><br></div><div>a) Downloaded VM Player and Ubuntu ISO</div><div><br></div><div><a href="http://theholmesoffice.com/installing-ubuntu-in-vmware-player-on-windows/" target="_blank">http://theholmesoffice.com/installing-ubuntu-in-vmware-player-on-windows/</a></div><div><br></div><div>b) installed Ubuntu 14 something on Windows 7</div><div><br></div><div>(By the way the Wubi which is supposed to download and install Ubuntu turned out to be a dud)</div><div><br></div><div>c) installed Asterisk 11 </div><div><br></div><div><a href="http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/" target="_blank">http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/</a></div><div><br></div><div>By the way, there was an undocumented problem with compiling DAHDI </div><div>So I skipped that step</div></div></div></blockquote><div><br></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline">​In a virtual environment, the DAHDI library is not useful, as it only serves to connect to hardware cards that you likely don't have and VMware generally doesn't support passing through to the virtual machine anyway.</div></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline"> ​</div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div dir="ltr"><div><br></div><div>d) ran Asterisk and everything is back where they should be</div><div><br></div><div>However, some times I get "User Busy" response from Vonage. I think this</div><div>is an Asterisk issue. If anyone knows how to rejig Asterisk so that it won't hold</div><div>on to the session after hang up, kindly let me know.</div><div><br></div></div></div></blockquote><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline"><br></div></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline">To get assistance with specific SIP call failures, you would need to capture the SIP messaging for an instance where it failed, using 'sip set debug' in Asterisk or wireshark, and then share that.  You'll only need the actual SIP traffic on port 5060, not the RTP.</div></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline"><br></div></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline">I would also recommend testing with a different provider (ITSP) first.  I have used <a href="http://voip.ms">voip.ms</a> successfully with Asterisk.</div></div><div><div class="gmail_default" style="color:rgb(102,0,0);display:inline">​</div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div dir="ltr"><div></div><div>Best regards</div><span class="HOEnZb"><font color="#888888"><div>murthy</div>                                       </font></span></div></div>
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asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><img alt="Digium logo" src="https://my.digium.com/images/graphics/digium_RGB_signature.gif" width="288" height="50" style="color:rgb(0,0,0);font-family:Arial,Helvetica,sans-serif;font-size:12px"><div>Scott Griepentrog<br>Digium, Inc · Software Developer<br>445 Jan Davis Drive NW · Huntsville, AL 35806 · US<br>direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090<br>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> · <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br></div></div></div>
</div></div>