title: Asterisk 13.7.2 pjsip 2.4.5 unexpected BYE with SIP cause 58 while answering
body: <p>I'm upgrading Asterisk 11 to 13 and testing new configuration with WebRTC enpoints. For some reason PJSIP is dropping call sending BYE to replying endpoint after receiving OK packet from that end. No any errors in the console, only  <em>== Everyone is busy/congested at this time (1:0/0/1)</em> populated between sending ACK and BYE </p>
  
  <p>At the same time clients are able to place a call and listen messages or perform calls routed via iax</p>
  
  <p>With same scenario Asterisk 11 just establishing connection:</p>
  
  <pre><code>SIP/920-00000009 answered SIP/929-00000008
  Audio is at 10292
  Adding codec 100003 (ulaw) to SDP
  </code></pre>
  
  <p>Could you please advise what could be a reason for such behaviour?
  P.S with the same config Asterisk 13 is sending/receiving SIP MESSAGE perfectly</p>
  
  <p>Because this forum is not happy to populate SIP debug output as code, you can refer my <a href="https://community.asterisk.org/t/asterisk-13-7-2-pjsip-2-4-5-unexpected-bye-with-sip-cause-58-while-answering/66066?u=dimagolov" rel="nofollow">post on the same</a> at Asterisk community forum for SIP debug, pjsip.conf endpoint configuration and pjsip show contacts output</p>