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    <div class="moz-cite-prefix">Hello,<br>
      <br>
      What OpenSIPS version are you currently using ?<br>
      I've just committed a fix that implements a preliminary version of
      this, see commits :<br>
      <br>
<a class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48">https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48</a><br>
      and<br>
<a class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33">https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33</a><br>
      <br>
      Please apply them to your sources and let me know how it oges<br>
      Best Regards,<br>
      <pre class="moz-signature" cols="72">Vlad Paiu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a> </pre>
      On 25.08.2014 13:31, Satish Patel wrote:<br>
    </div>
    <blockquote
      cite="mid:637B9FCF-B53C-4793-B34C-C1B4C23A82C6@gmail.com"
      type="cite">
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      <div>Great!! I can see light in tunnel now because last 1 week I
        tried everything and now I was planing to go for B2B but I guess
        as you said you guys working on so I'm holding my breath.&nbsp;</div>
      <div><br>
      </div>
      <div>This is must needed solution because SIP service provide most
        of time provide password to make outbound trunk call.&nbsp;<br>
        <br>
        Sent from my iPhone</div>
      <div><br>
        On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu &lt;<a
          moz-do-not-send="true" href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>&gt;
        wrote:<br>
        <br>
      </div>
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          <div class="moz-cite-prefix"><tt>Hi Satish,<br>
              <br>
              It is an known issue that OpenSIPS does not increases the
              cseq number when performing UAC auth against another
              party. Asterisk does not like that and consider the new
              branch INVITE with credentials a simple retransmission
              (even if it has a different VIA-branch :P) and discards
              them - this is why you get that timeout from asterisk.<br>
              <br>
              We have ongoing work (hopefully to be ready in 1-2 weeks)
              for increasing the cseq number is a sip-wise manner. Just
              keep an eye on the mailing list.<br>
              <br>
              Regards,<br>
            </tt>
            <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
            On 25.08.2014 04:47, Satish Patel wrote:<br>
          </div>
          <blockquote
cite="mid:CAPgF-frYOOyynD-7fH1RWUYgLizSfU4yuuimPJE0+owTrLU6pA@mail.gmail.com"
            type="cite">
            <div dir="ltr">
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                <div>
                  <div>
                    <div>
                      <div>
                        <div>
                          <div>
                            <div>
                              <div>I am seeing following and all
                                transaction has CSeq: 2 INVITE, I have
                                notice one thing asterisk asking for 407
                                but opensips never send any challenge
                                response <br>
                                <br>
                              </div>
                              Opensips ---&gt; INVITE ---&gt; Asterisk<br>
                            </div>
                            Asterisk -----&gt; 407 ------&gt; Opensips&nbsp;&nbsp;
                            (Proxy-Authenticate: Digest algorithm=MD5,
                            realm="asterisk", nonce="3a710e79".)<br>
                          </div>
                          Opensips ----&gt; ACK -----&gt; Asterisk<br>
                        </div>
                        <br>
                      </div>
                      Here opensips challenging SIP client&nbsp; and saying
                      giving try to asterisk and then following<br>
                      <br>
                    </div>
                    Opensips ----&gt; INVITE ---&gt; Asterisk<br>
                  </div>
                  Opensips ----&gt; INVITE ----&gt; Asterisk <br>
                </div>
                Opensips ----&gt; INVITE ----&gt; Asterisk <br>
                <br>
              </div>
              After 3 tries opensips send SIP client 408 Request
              timeout.. <br>
            </div>
            <div class="gmail_extra"><br>
              <br>
              <div class="gmail_quote">On Sun, Aug 24, 2014 at 4:26 PM,
                Stefano Pisani <span dir="ltr">&lt;<a
                    moz-do-not-send="true"
                    href="mailto:stefano.pisani@omnianet.it"
                    target="_blank">stefano.pisani@omnianet.it</a>&gt;</span>
                wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  <div text="#000000" bgcolor="#FFFFFF">
                    <div>Check if the cseq was incremented by one in the
                      second try.<br>
                      Use ngrep.<br>
                      <br>
                      <br>
                      <br>
                      Il 24/08/2014 22.24, Satish Patel ha scritto:<br>
                    </div>
                    <blockquote type="cite">
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                                  <div><br>
                                  </div>
                                  Hi,<br>
                                  <br>
                                </div>
                                my Opensips (UAC) registered to PSTN
                                gateway and now i am trying to call
                                using my SIPphone which is register to
                                opensip but no success. I am getting 407
                                Proxy authentication issue..&nbsp; I am using
                                following method but it didn't work. I
                                need solution badly.. <br>
                                <br>
                              </div>
                              PSTN gateway sending 407 Proxy auth and
                              then my Opensip sending 407 proxy auth to
                              SIP phone.&nbsp; <br>
                              <br>
                            </div>
                            Does anyone has any working example or some
                            kind of document? I haven;t see any single
                            doc anywhere in Internet about uac_auth&nbsp;
                            issue<br>
                            <div>
                              <div>
                                <div><br>
                                  <br>
                                  <br>
                                  <pre>modparam("uac","credential","username:domain:password")

route {
....
            t_on_failure("2");
            t_relay( "udp:ip_addr:5060" );
...
}

failure_route[2] {
     uac_auth();
     t_relay("udp:ip_addr:5060");
}
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                      <pre>_______________________________________________
Users mailing list
<a moz-do-not-send="true" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a moz-do-not-send="true" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
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                    <br>
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                  <br>
                  _______________________________________________<br>
                  Users mailing list<br>
                  <a moz-do-not-send="true"
                    href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
                  <a moz-do-not-send="true"
                    href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users"
                    target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
                  <br>
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            <pre wrap="">_______________________________________________
Users mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
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      <pre wrap="">_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
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