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    Hi Joegen,<br>
    <br>
    I just wanted to say a quick thank you for your advice - it worked.
    As soon as I turned off all the NAT related stuff on the Aastra
    phone (basically turned off rport and removed the stun and turn
    servers), the phone stopped changing its VIA header and CANCEL
    worked as advertised.<br>
    <br>
    In case it's useful to anyone else in a similar position, I have the
    following NAT related stuff enabled on the internal FS sip profile
    (note: my FS server is on a public IP i.e. not itself behind NAT):<br>
    <br>
    &lt;param name="NDLB-received-in-nat-reg-contact" value="true"/&gt;<br>
    &lt;param name="NDLB-force-rport" value="true"/&gt;<br>
    &lt;param name="nat-options-ping" value="true"/&gt;<br>
    <br>
    Charlie<br>
    <br>
    On 03/12/2011 12:37, Charlie Orford wrote:
    <blockquote cite="mid:4EDA09FC.9000504@attackplan.net" type="cite">
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      Bummer - we've got an office full of these phones :(<br>
      <br>
      As you suggest, I'll turn off all the NAT stuff when I'm there on
      Monday and see if that fixes it - hope so, otherwise I guess it
      looks like I'll have to try and get Aastra to look at the problem
      (doubt they are going to be very interested though!).<br>
      <br>
      Thanks,<br>
      Charlie<br>
      <br>
      <br>
      On 03/12/2011 06:29, Joegen Baclor wrote:
      <blockquote cite="mid:4ED9B3A8.4040707@ezuce.com" type="cite">
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        It's definitely a bug on the phone.&nbsp; I would try setting all
        Aastra NAT related features off and see if it stops getting
        confused like this.&nbsp; Freeswitch should be able to handle NAT
        from the far end. <br>
        <br>
        On 12/02/2011 08:18 PM, Charlie Orford wrote:
        <blockquote cite="mid:4ED8C206.5010502@attackplan.net"
          type="cite">
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          Thanks Joegen, I hadn't picked up on that.<br>
          <br>
          Is there any valid reason why the phone would swap to using
          it's internal IP (192.168.0.175) for the Via and Contact
          headers when responding to the auth challenge? Or does this
          look more like a bug with the phone?<br>
          <br>
          <br>
          <br>
          On 02/12/2011 12:50, Joegen Baclor wrote:
          <blockquote cite="mid:4ED8BB73.8080401@ezuce.com" type="cite">
            For some reason, the aastra phone is sending a different
            contact&nbsp; and via address after it gets challenged by
            freeswitch.<br>
            <br>
            Original address was <span><br>
              <br>
            </span> <span>Via: SIP/</span><span>2.0</span><span>/UDP<span>&nbsp;</span></span><span>7.7</span><span>.7</span><span>.7</span><span>:</span><span>6060</span><span>;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport</span>&nbsp;&nbsp;




            (LINE# 35)<br>
            <br>
            Then it changes to <br>
            <br>
            <span>Via: SIP/</span><span>2.0</span><span>/UDP<span>&nbsp;</span></span><span>192.168</span><span>.0</span><span>.175</span><span>:</span><span>6060</span><span>;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport</span>
            (LINE# 187)<br>
            <br>
            It then sends a CANCEL using <span>Via: SIP/</span><span>2.0</span><span>/UDP<span>&nbsp;</span></span><span>7.7</span><span>.7</span><span>.7</span><span>:</span><span>6060


              (LINE# 961) which freeswitch isn't able to match to a
              transaction because it is probably expecting </span><span><span></span></span><span>192.168</span><span>.0</span><span>.175.<br>
              <br>
              <br>
            </span><br>
            On 12/02/2011 07:19 PM, Charlie Orford wrote:
            <blockquote cite="mid:4ED8B456.9000505@attackplan.net"
              type="cite"> Hi Anthony<br>
              <br>
              We are using Aastra 57i phones with the latest firmware
              (v3.2.2.56 from June 2011).<br>
              <br>
              However, it looks like a specific issue with the Aastra
              57i as I have just replicated the call and cancel
              situation using a 3CX softphone and in this case the
              CANCEL request gets honoured by FS.<br>
              <br>
              Here's a copy of a failed CANCEL test using the Aastra
              phone: <a moz-do-not-send="true"
                href="http://pastebin.freeswitch.org/17922">http://pastebin.freeswitch.org/17922</a><br>
              <br>
              Here's a copy of a successful CANCEL test using the 3CX
              softphone: <a moz-do-not-send="true"
                href="http://pastebin.freeswitch.org/17923">http://pastebin.freeswitch.org/17923</a><br>
              <br>
              sofia loglevel all 9 is on for both tests.<br>
              <br>
              Charlie<br>
              <br>
              <br>
              On 01/12/2011 21:53, Anthony Minessale wrote:
              <blockquote
cite="mid:CAKbxfG_UrV0q=+Pu_ja+kftrvuuB-FN3oKiG6H08_uUpDr2CGQ@mail.gmail.com"
                type="cite">try same failed call test with sofia
                loglevel all 9
                <div><br>
                </div>
                <div>Also, try some other phone or device that does not
                  have the problem, and create a cancel situation the
                  same way and see if you can find a difference.</div>
                <div><br>
                </div>
                <div>Finally, have you tried the latest firmware on the
                  phone?</div>
                <div><br>
                  <br>
                  <div>On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford <span>&lt;<a
                        moz-do-not-send="true"
                        href="mailto:charlie.orford@attackplan.net">charlie.orford@attackplan.net</a>&gt;</span>
                    wrote:<br>
                    <blockquote>Hi list,<br>
                      <br>
                      When we make a call from an FS extension to a PSTN
                      number (via our ITSP<br>
                      gateway provider) and hang-up before the call
                      completes, FS replies to<br>
                      the CANCEL request with "481 Call/Transaction Does
                      Not Exist" and the<br>
                      call continues to ring on the remote end. If we
                      hangup after the call<br>
                      has connected, it works with no problem.<br>
                      <br>
                      I have looked through a SIP trace of this
                      happening and to my (untrained<br>
                      eye) nothing seems obviously wrong (i.e. tag, call
                      id and branch values<br>
                      all seem to be correct). I'm using the latest git
                      snapshot from<br>
                      2011-11-30 18-14-24 -0600.<br>
                      <br>
                      For a sip trace showing the problem, see:<br>
                      <a moz-do-not-send="true"
                        href="http://pastebin.freeswitch.org/17908">http://pastebin.freeswitch.org/17908</a><br>
                      <br>
                      For a sip trace showing a successful hangup, see:<br>
                      <a moz-do-not-send="true"
                        href="http://pastebin.freeswitch.org/17909">http://pastebin.freeswitch.org/17909</a><br>
                      <br>
                      *Note: ip addresses, domain and called number have
                      been altered for privacy.<br>
                      <br>
                      Any help or insight is much appreciated.<br>
                      <br>
                      Kind Regards,<br>
                      Charlie<br>
                      <br>
                      <br>
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                  <br>
                  <br>
                  <div><br>
                  </div>
                  -- <br>
                  Anthony Minessale II<br>
                  <br>
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      <br>
      <pre wrap="">_________________________________________________________________________
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