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    Hello,<br>
    <br>
    you must configure asterisk to trust the traffic coming from
    Kamailio IP and not authenticate.<br>
    <br>
    Cheers,<br>
    Daniel<br>
    <br>
    On 8/29/10 5:00 PM, <a class="moz-txt-link-abbreviated" href="mailto:chuks@cybernergies.com">chuks@cybernergies.com</a> wrote:
    <blockquote
cite="mid:20100829080044.9cebde9d7766d7119f10221126f9914a.457d4c18a0.wbe@email00.secureserver.net"
      type="cite"><span style="font-family: Verdana; color: rgb(0, 0,
        0); font-size: 10pt;">
        <blockquote id="replyBlockquote" webmail="1" style="border-left:
          2px solid blue; margin-left: 8px; padding-left: 8px;
          font-size: 10pt; color: black; font-family: verdana;">
          <div id="wmQuoteWrapper"><span style="font-family: Verdana;
              color: rgb(0, 0, 0); font-size: 10pt;">
              <div>Hello<br>
                 I have been having issues with kamailio and asterisk
                realtime. I have used all the configurations posted, but
                it just has not worked for me. What I am trying to do is
                to use asterisk as PSTN &amp; voicemail for kamailio.
                But I keep getting this 401 not authorized from asterisk
                like this:</div>
              <div><br>
              </div>
              <div>========</div>
              <div>
                <div>--- (19 headers 19 lines) ---</div>
                <div>Sending to 99.89.26.17:5060 (NAT)</div>
                <div>Using INVITE request as basis request -
                  LCklNT_HoIeTxCB_8cSIf9efRNvkcR</div>
                <div>       &gt; doing dnsmgr_lookup for '99.89.26.17'</div>
                <div>    -- adding dns manager for '99.89.26.17'</div>
                <div>Scheduling destruction of SIP dialog
                  '<a class="moz-txt-link-abbreviated" href="mailto:3d67147a652fe8641c536eb92383ad74@99.89.26.17">3d67147a652fe8641c536eb92383ad74@99.89.26.17</a>' in
                  32000 ms (Method: NOTIFY)</div>
                <div>Reliably Transmitting (NAT) to 99.89.26.17:5060:</div>
                <div>NOTIFY <a class="moz-txt-link-freetext" href="sip:1000@99.89.26.17">sip:1000@99.89.26.17</a> SIP/2.0</div>
                <div>Via: SIP/2.0/UDP
                  99.89.26.18:5060;branch=z9hG4bK00545e8e;rport</div>
                <div>Max-Forwards: 70</div>
                <div>From: "asterisk"
                  <a class="moz-txt-link-rfc2396E" href="sip:1000@99.89.26.17">&lt;sip:1000@99.89.26.17&gt;</a>;tag=as4d28adb7</div>
                <div>To: <a class="moz-txt-link-rfc2396E" href="sip:1000@99.89.26.17">&lt;sip:1000@99.89.26.17&gt;</a></div>
                <div>Contact: <a class="moz-txt-link-rfc2396E" href="sip:1000@99.89.26.18:5060">&lt;sip:1000@99.89.26.18:5060&gt;</a></div>
                <div>Call-ID:
                  <a class="moz-txt-link-abbreviated" href="mailto:3d67147a652fe8641c536eb92383ad74@99.89.26.17">3d67147a652fe8641c536eb92383ad74@99.89.26.17</a></div>
                <div>CSeq: 102 NOTIFY</div>
                <div>User-Agent: Asterisk PBX</div>
                <div>Event: message-summary</div>
                <div>Content-Type: application/simple-message-summary</div>
                <div>Content-Length: 84</div>
                <div><br>
                </div>
                <div>Messages-Waiting: no</div>
                <div>Message-Account: <a class="moz-txt-link-freetext" href="sip:1@99.89.26.17">sip:1@99.89.26.17</a></div>
                <div>Voice-Message: 0/0 (0/0)</div>
                <div><br>
                </div>
                <div>---</div>
                <div>Found peer '1000' for '1000' from 99.89.26.17:5060</div>
                <div><br>
                </div>
                <div>&lt;--- Reliably Transmitting (NAT) to
                  99.89.26.17:5060 ---&gt;</div>
                <div>SIP/2.0 401 Unauthorized</div>
                <div>Via: SIP/2.0/UDP
99.89.26.17;branch=z9hG4bK354f.e07b39b2.0;received=99.89.26.17;rport=5060</div>
                <div>Via: SIP/2.0/UDP
                  192.168.1.101:5060;branch=z9hG4bKj4sndhbgkh863bu8fpr61tb;rport=5060</div>
                <div>From:
                  <a class="moz-txt-link-rfc2396E" href="sip:1000@99.89.26.17">&lt;sip:1000@99.89.26.17&gt;</a>;tag=5tnt79v6phhc689kd5vh</div>
                <div>To:
                  <a class="moz-txt-link-rfc2396E" href="sip:+2348023098407@99.89.26.17;user=phone">&lt;sip:+2348023098407@99.89.26.17;user=phone&gt;</a>;tag=as21d7a164</div>
                <div>Call-ID: LCklNT_HoIeTxCB_8cSIf9efRNvkcR</div>
                <div>CSeq: 1710 INVITE</div>
                <div>Server: Asterisk PBX</div>
                <div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
                  SUBSCRIBE, NOTIFY, INFO, PUBLISH</div>
                <div>Supported: replaces, timer</div>
                <div>WWW-Authenticate: Digest algorithm=MD5,
                  realm="99.89.26.17", nonce="16c11ac7"</div>
                <div>Content-Length: 0</div>
                <div style=""><br style="">
                </div>
                <div>========</div>
                <div><br>
                </div>
                <div>asterisk and kamailio are on different server, and
                  I have put the IP of asterisk in trusted table in
                  kamailio db. My kamailio.cfg is:</div>
                <div><br>
                </div>
                <br>
              </div>
            </span></div>
        </blockquote>
      </span></blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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