Hi, the tcpdump starts with a pretty standard INVITE sequence:
10.0.0.121 -- INVITE --> 10.0.0.3 10.0.0.121 <-- 401 Unauthorized -- 10.0.0.3 // asterisk gives nonce in WWW-Authenticate: header 10.0.0.121 -- ACK --> 10.0.0.3 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. 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? Can you provide the same logs for a successful call? Do incoming calls to the phone work when this happens? -- BR, marie On 02.05.2018, at 20:23, John Kinsner <[email protected]> wrote: > 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. > > 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. > > sip debug from asterisk: > https://pastebin.com/Mmz9JsAP > > tcpdump from pbx: > https://pastebin.com/jRT9QJwq > > sip.conf: > [121] > type=friend > ;qualify=yes > ;qualifyfreq=300 > host=dynamic > context=extensions > secret=MySecret > nat=no > callerid="MBR" <121> > > > can anyone give me clues to troubleshoot? > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
