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    <div class="moz-cite-prefix">You will have to do IP auth on the
      Kamailio coming from Asterisk<br>
      <br>
      <br>
      On 6/17/13 4:07 PM, Mickael MONSIEUR wrote:<br>
    </div>
    <blockquote
cite="mid:CAK2CCpeBRMnSMcdXGCQHp=Ps-0dZtqtwMEa8S8gUQYvO-j0Jzw@mail.gmail.com"
      type="cite">
      <div dir="ltr">Hello,
        <div style="">First, i have 7 years experience with Asterisk,
          but I started a project with Kamailio, forgive me in advance
          if I say silly things...! ;-)</div>
        <div style=""><br>
        </div>
        <div style="">
          <div>I set up a classic Asterisk / Kamailio configuration:</div>
          <div>sip phones -> kamailio -> asterisk -> sip
            trunks/pstn.</div>
          <div><br>
          </div>
          <div>
            <div>When a call comes from the PSTN side, if I configure
              Asterisk as follows:</div>
            <div><br>
            </div>
            <div>
              <div>
                <div>[012345678]</div>
                <div>type = friend</div>
                <div>username = 012345678</div>
                <div>secret = xxxxxxx</div>
                <div>host = dynamic</div>
              </div>
              <div>fromdomain = <a moz-do-not-send="true"
                  href="http://sip.mydomain.com">sip.mydomain.com</a></div>
              <div>fromuser = 012345678</div>
            </div>
            <div><br>
            </div>
            <div>Standard mode:</div>
            <div>exten => 012345678, 1, Dial(SIP/012345678) -> The
              call is redirected on the phone by Kamailio ! :-)</div>
            <div><br>
            </div>
            <div>------------------------------------------------------------------------------------------------------------------------------------------------<br>
            </div>
            <div><br>
            </div>
            <div><br>
            </div>
            <div>Trunk mode:</div>
            <div><br>
            </div>
            <div>
              <div>
                <div>[mytrunk]</div>
                <div>type = friend</div>
                <div>username = mytrunkUser</div>
                <div>secret = xxxxxxx</div>
                <div>host = dynamic</div>
              </div>
              <div>
                fromdomain = <a moz-do-not-send="true"
                  href="http://sip.mydomain.com">sip.mydomain.com</a></div>
              <div>fromuser = mytrunkUser</div>
            </div>
            <div><br>
            </div>
            <div><br>
            </div>
            <div>exten => 012345678, 1, Dial(SIP/mytrunk/012345678)
              -> The call is rejected by Kamailio....</div>
            <div>exten => 012345679, 1, Dial(SIP/mytrunk/012345679)
              -> The call is rejected by Kamailio ....</div>
          </div>
          <div><br>
          </div>
          <div><br>
          </div>
          <div>My question is how to allow the routing of multiple
            numbers (trunk mode) in a SIP account with Kamailio?<br>
          </div>
          <div style="">Best regards,</div>
          <div style="">Mickael</div>
          <div><br>
          </div>
        </div>
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      <pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
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