So you have executed this call scenario: 1-a, 2-a, 3-a, 1-b, 2-b, 3-b, 1-c, 2-c, 3-c and got failure on each of the 9 calls? And then replicated on the "good" call (1-a,2-a.)?
_____ From: [email protected] [mailto:[email protected]] On Behalf Of Ott Rose Sent: Wednesday, August 12, 2009 11:08 AM To: [email protected] Subject: Re: [asterisk-users] call drops after a few seconds we have three phones hooked up right now. we have tried on all the different phones and have called several different external numbers. all with the same result. > From: [email protected] > To: [email protected] > Date: Wed, 12 Aug 2009 10:48:32 -0500 > Subject: Re: [asterisk-users] call drops after a few seconds > > Have you tried to "replicate" the problem (call from a to b 3-5 consecutive > times to see if same result)? > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Ishfaq Malik > Sent: Wednesday, August 12, 2009 10:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] call drops after a few seconds > > 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 > > HotmailR. 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 > > > _______________________________________________ > -- 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 _____ Get back to school stuff for them and cashback for you. Try Bing now. <http://www.bing.com/cashback?form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_Back ToSchool_Cashback_BTSCashback_1x1>
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