<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">Hi,<div><br></div><div>the tcpdump starts with a pretty standard INVITE sequence:</div><div><br></div><div>10.0.0.121 -- INVITE --> 10.0.0.3</div><div>10.0.0.121 <-- 401 Unauthorized -- 10.0.0.3<span class="Apple-tab-span" style="white-space:pre"> </span>// asterisk gives nonce in WWW-Authenticate: header</div><div>10.0.0.121 -- ACK --> 10.0.0.3</div><div><br></div><div>After that, normally you would see a new INVITE from the phone with Authorization: header, but in your case the phone does not send this - although it is clearly reachable as indicated by the SIP OPTIONS dialogue.</div><div><br></div><div>In the Asterisk SIP debug, I see only the packets sent by Asterisk to the phone, but not the phone's responses. Did you do just 'sip set debug on' or something different?</div><div><br></div><div>Can you provide the same logs for a successful call?</div><div><br></div><div>Do incoming calls to the phone work when this happens?</div><div><br></div><div>-- </div><div><div apple-content-edited="true"><div>BR,</div><div><br></div><div>marie</div>
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<br><div><div>On 02.05.2018, at 20:23, John Kinsner <<a href="mailto:jkinsner1@yahoo.com">jkinsner1@yahoo.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div><div style="font-family:Helvetica Neue, Helvetica, Arial, sans-serif;font-size:16px;"><div> <div><div><div><div>sometime during the past few upgrades on asterisk 13, my Cisco 7965G phones are sporadically not able to make calls. after a few seconds, they just play a fast-busy tone. I tried upgrading the 7965G OS from their original (working for years) 9.4.2SR1 to 9.4.2SR3 and the behavior did not change.<br><div><br>they are talking via chan_sip on asterisk 13.19.0. I cannot determine the sporadic part, sometimes the call goes through fine with no configuration changes or restarts/reboots on either end.<br></div></div><br></div>sip debug from asterisk:<br></div><a href="https://pastebin.com/Mmz9JsAP" rel="nofollow" target="_blank">https://pastebin.com/Mmz9JsAP</a></div><div><div><br><div>tcpdump from pbx:<br></div></div><a href="https://pastebin.com/jRT9QJwq" rel="nofollow" target="_blank" class="">https://pastebin.com/jRT9QJwq</a></div><div><br><div>sip.conf:<br><span>[121]<br>type=friend<br>;qualify=yes<br>;qualifyfreq=300<br>host=dynamic<br>context=extensions<br>secret=MySecret<br>nat=no<br>callerid="MBR" <121></span><br></div></div><div><br><div><br><div>can anyone give me clues to troubleshoot?<br><br></div></div></div><div><br></div></div><div><br></div><div><br></div><div><br></div></div></div>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div></body></html>