I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn&#39;t say that because it seems like Nat is not a very basic setup with Asterisk.<div>
<br></div><div>The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with <a href="http://voip.ms">voip.ms</a>. </div>
<div><br></div><div>But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :)</div><div><br></div><div>Thanks</div><div><br><br><div class="gmail_quote">On Mon, Sep 12, 2011 at 11:09 AM, John Novack <span dir="ltr">&lt;<a href="mailto:jnovack@stromberg-carlson.org">jnovack@stromberg-carlson.org</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    Never have had a problem with their IAX service.<br>
    <br>
    And ( for now ) a little hedge against the hackers.<br>
    <br>
    Since Asterisk is involved, why not use IAX anyway?<br><font color="#888888">
    <br>
    <br>
    John Novack</font><div><div></div><div class="h5"><br>
    <br>
    <br>
    naren wrote:
    <blockquote type="cite">
      <div><br>
      </div>
      <div>I also found this... seems like <a href="http://voip.ms" target="_blank">voip.ms</a> outbound is broken for now!</div>
      <div><br>
      </div>
      <div><a href="http://pbxinaflash.com/forum/showthread.php?t=10735" target="_blank">http://pbxinaflash.com/forum/showthread.php?t=10735</a></div>
      <div><br>
      </div>
      <br>
      <br>
      <div class="gmail_quote">On Sun, Sep 11, 2011 at 10:34 PM, naren <span dir="ltr">&lt;<a href="mailto:naren.salem@gmail.com" target="_blank">naren.salem@gmail.com</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
          Hi,
          <div><br>
          </div>
          <div>I am trying to set up my asterisk 1.8.5 with <a href="http://voip.ms" target="_blank">voip.ms</a>. I had no problem with the
            incoming, but my outgoing is not working. If at all
            possible, I would like to stick with SIP. Since the original
            poster (Glen) had mentioned that he had gotten outgoing
            working, I was wondering if you would be kind enough to post
            some thoughts on that. Were you able to get it working with
            just the default example sip.conf / extensions.conf settings
            that they have on their website?</div>
          <div><br>
          </div>
          <div>I have pretty much the same settings. When I dial out,
            the destination rings, but I can&#39;t hear a ringback tone from
            on the source side ( I am using a PAP2T router with a
            phone). I have set up outgoing with actionvoip before and
            that is working fine, so I am thinking my router settings
            for my ports are correct - but I am no expert.</div>
          <div><br>
          </div>
          <div>I would really appreciate it if you could post the
            relevant section of your sip.conf for me.</div>
          <div><br>
          </div>
          <div>Thanks!</div>
          <div>Naren</div>
          <div>
            <div>
              <div><br>
                <br>
                <div class="gmail_quote">
                  On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <span dir="ltr">&lt;<a href="mailto:asterisk.org@sedwards.com" target="_blank">asterisk.org@sedwards.com</a>&gt;</span>
                  wrote:<br>
                  <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                    <div>On Thu, 9 Jun 2011, John Novack wrote:<br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        I use <a href="http://voip.ms" target="_blank">voip.ms</a>
                        and have no issues using IAX and Asterisk 1.4.xx<br>
                      </blockquote>
                      <br>
                    </div>
                    &#39;slam-dunk.&#39;
                    <div><br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        Though they suggest SIP, I chose IAX and have
                        4569 UDP open in my firewall<br>
                      </blockquote>
                      <br>
                      a<br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        Their on line config samples just work!<br>
                      </blockquote>
                      <br>
                    </div>
                    is
                    <div><br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        Suggest you check your firewall and your
                        configs, and above all post some more
                        information<br>
                      </blockquote>
                      <br>
                    </div>
                    IAX
                    <div><br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        If you really want to upset some, top post as I
                        have just done!<br>
                      </blockquote>
                      <br>
                    </div>
                    Agreed.
                    <div><br>
                      <br>
                      <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                        The real issue is communication, top bottom or
                        in the middle<br>
                      </blockquote>
                      <br>
                    </div>
                    Sometimes, it&#39;s just about being considerate to &#39;the
                    next guy.&#39;<br>
                    <font color="#888888">
                      <br>
                      -- <br>
                      Thanks in advance,<br>
                      -------------------------------------------------------------------------<br>
                      Steve Edwards       <a href="mailto:sedwards@sedwards.com" target="_blank">sedwards@sedwards.com</a>    
                       Voice: <a href="tel:%2B1-760-468-3867" value="+17604683867" target="_blank">+1-760-468-3867</a>
                      PST<br>
                      Newline                                          
                         Fax: <a href="tel:%2B1-760-731-3000" value="+17607313000" target="_blank">+1-760-731-3000</a></font>
                    <div>
                      <div><br>
                        <br>
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                    </div>
                  </blockquote>
                </div>
                <br>
              </div>
            </div>
          </div>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>--
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    </blockquote>
    <br>
    </div></div><div class="im"><pre cols="10000">-- 

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

</blockquote></div><br></div>