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    <p><font face="Cantarell">Hello Alexey,</font></p>
    <p><font face="Cantarell">Thank you on reply,</font></p>
    <p><font face="Cantarell">I undone all changes regard headers
        changes and MS Teams send BYE directly to asterisk.<br>
      </font></p>
    <p><font face="Cantarell">No Route header present.</font></p>
    <p><font face="Cantarell">But INVITE ACK 183 180 all travel with 
        proper routing information.<br>
      </font></p>
    <p><font face="Cantarell"><br>
      </font></p>
    <p><font face="Cantarell">2020/04/18 17:54:28.599711
        190.109.70.77:5060 -> 190.109.68.250:5060<br>
        BYE <a class="moz-txt-link-freetext" href="sip:11988582770@190.109.68.250:5060">sip:11988582770@190.109.68.250:5060</a> SIP/2.0<br>
        FROM:
<a class="moz-txt-link-rfc2396E" href="sip:551120999838@190.109.70.77"><sip:551120999838@190.109.70.77></a>;tag=4d7fb0763c224e39a13a03c669c4b387<br>
        TO: <a class="moz-txt-link-rfc2396E" href="sip:11988582770@190.109.70.77"><sip:11988582770@190.109.70.77></a>;tag=as41e97ff5<br>
        CSEQ: 3 BYE<br>
        CALL-ID: <a class="moz-txt-link-abbreviated" href="mailto:2e6c1a8d2383a4752403e94512ced077@190.109.70.77">2e6c1a8d2383a4752403e94512ced077@190.109.70.77</a><br>
        MAX-FORWARDS: 69<br>
        Via: SIP/2.0/UDP
        190.109.70.77:5060;branch=z9hG4bK050e.e400e373.0;i=66c9c603<br>
        VIA: SIP/2.0/TLS
        52.114.14.70:5061;rport=8208;received=52.114.14.70;branch=z9hG4bK9594cd7<br>
        REASON:
        Q.850;cause=18;text="fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;Call
        Controller timed out while waiting for acknowledgement."<br>
        CONTACT:
<a class="moz-txt-link-rfc2396E" href="sip:52.114.14.70:8208;nat=yes;x-i=fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;x-c=b5841f98785c5819bb99e57cd0fa7d86/s/1/9b6ba2f8eefa4a67bed29609fd1884ec"><sip:52.114.14.70:8208;nat=yes;x-i=fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;x-c=b5841f98785c5819bb99e57cd0fa7d86/s/1/9b6ba2f8eefa4a67bed29609fd1884ec></a><br>
        CONTENT-LENGTH: 0<br>
        USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.13.7 i.ASSE.3<br>
        ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY<br>
      </font></p>
    <p><font face="Cantarell"><br>
      </font></p>
    <p><font face="Cantarell">volga629<br>
      </font></p>
    <p><font face="Cantarell"></font><br>
    </p>
    <div class="moz-cite-prefix">On 4/18/20 5:13 PM, Alexey Vasilyev
      wrote:<br>
    </div>
    <blockquote type="cite"
      cite="mid:1587240834447-0.post@n2.nabble.com">
      <pre class="moz-quote-pre" wrap="">Hi volga629,

There were nothing special for ACK. You don't need to change
To/From/Contact. All the necessary steps were in the article
<a class="moz-txt-link-freetext" href="https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/">https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/</a> and for most
people it still works.
So I'm not sure, that MS changed anything, because all the hardware SBCs
should change behaviour, so they need new firmware. SBC vendors should
inform customers to update etc. So this is not so simple process. And it
definitely make no sense for anybody.
And in the test lab for the article I've used absolutely the same
architecture with asterisk, the only difference was RTPEngine to transcode
SRTP-RTP.
And within test lab I've tested not only calls, but transfers worked fine
too.



-----
---
Alexey Vasilyev
--
Sent from: <a class="moz-txt-link-freetext" href="http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html">http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html</a>

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</pre>
    </blockquote>
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