I've encountered this issue a couple of times and we managed to resolve it by updating the sip phone and the router it was connected to both to use their latest firmware.
I know it's not a definitive answer but I've never truly got down to the heart of the issue as with us it would affect just one out of 100 or so extensions. Ish Ott Rose wrote: > I have setup my asterisk box using freepbx. I can call extension and > make outbound calls. the outbound calls drop between 10-30sec. we are > using bandwidth.com and they have logged our call. below is your bad > followed by what they say is a good call. I can't figure out where the > problem is on your end. I know we are missing some stuff at the bottom > but I don't know where to start. > > **************BAD CALL************************ > Wed Aug 5 18:22:28 2009 64.191.130.78:5060 ---> 216.82.224.202:5060 > > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK20dc2d74;rport > From:"Justin's Face"<sip:[email protected]>;tag=as5d2a3b2a > To:<sip:[email protected]> > Contact:<sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 05 Aug 2009 18:22:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 230 > > > > ***********GOOD CALL*************************** > INVITE sip:[email protected]:5060 SIP/2.0 > Record-Route:<sip:4.79.212.229;lr;ftag=VPSF506071629460> > Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK8ec1.70782da5.0 > Via: SIP/2.0/UDP > 4.68.250.148:5060;branch=z9hG4bK506071629460-1246361886000 > From:"HIX > INC"<sip:[email protected];isup-oli=0>;tag=VPSF506071629460 > To:<sip:[email protected]:5060> > Call-ID: [email protected] > CSeq: 1 INVITE > Contact:<sip:[email protected]:5060;transport=udp> > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 173 > > v=0 > o=- 1249498358 1249498359 IN IP4 63.215.29.149 > s=- > c=IN IP4 63.215.29.149 > t=0 0 > m=audio 61030 RTP/AVP 0 18 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > > ------------------------------------------------------------------------ > Express your personality in color! Preview and select themes for > Hotmail®. Try it now. > <http://www.windowslive-hotmail.com/LearnMore/personalize.aspx?ocid=PID23391::T:WLMTAGL:ON:WL:en-US:WM_HYGN_express:082009> > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
