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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.<br><br>> From: danny@debsinc.com<br>> To: asterisk-users@lists.digium.com<br>> Date: Wed, 12 Aug 2009 10:48:32 -0500<br>> Subject: Re: [asterisk-users] call drops after a few seconds<br>> <br>> Have you tried to "replicate" the problem (call from a to b 3-5 consecutive<br>> times to see if same result)?<br>> <br>> -----Original Message-----<br>> From: asterisk-users-bounces@lists.digium.com<br>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ishfaq Malik<br>> Sent: Wednesday, August 12, 2009 10:34 AM<br>> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>> Subject: Re: [asterisk-users] call drops after a few seconds<br>> <br>> I've encountered this issue a couple of times and we managed to resolve <br>> it by updating the sip phone and the router it was connected to both to <br>> use their latest firmware.<br>> <br>> I know it's not a definitive answer but I've never truly got down to the <br>> heart of the issue as with us it would affect just one out of 100 or so <br>> extensions.<br>> <br>> Ish<br>> <br>> Ott Rose wrote:<br>> > I have setup my asterisk box using freepbx. I can call extension and <br>> > make outbound calls. the outbound calls drop between 10-30sec. we are <br>> > using bandwidth.com and they have logged our call. below is your bad <br>> > followed by what they say is a good call. I can't figure out where the <br>> > problem is on your end. I know we are missing some stuff at the bottom <br>> > but I don't know where to start.<br>> ><br>> > **************BAD CALL************************<br>> > Wed Aug 5 18:22:28 2009 64.191.130.78:5060 ---> 216.82.224.202:5060<br>> ><br>> > INVITE sip:+18599484787@216.82.224.202 SIP/2.0<br>> > Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK20dc2d74;rport<br>> > From:"Justin's Face"<sip:200@64.191.130.78>;tag=as5d2a3b2a<br>> > To:<sip:+18599484787@216.82.224.202><br>> > Contact:<sip:200@64.191.130.78><br>> > Call-ID: 3ffa6df00137d1923c69ca105bb3d091@10.0.0.8<br>> > CSeq: 102 INVITE<br>> > User-Agent: Asterisk PBX<br>> > Max-Forwards: 70<br>> > Date: Wed, 05 Aug 2009 18:22:28 GMT<br>> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> > Supported: replaces<br>> > Content-Type: application/sdp<br>> > Content-Length: 230<br>> ><br>> ><br>> ><br>> > ***********GOOD CALL***************************<br>> > INVITE sip:+19194393536@216.82.224.202:5060 SIP/2.0 <br>> > Record-Route:<sip:4.79.212.229;lr;ftag=VPSF506071629460><br>> > Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK8ec1.70782da5.0<br>> > Via: SIP/2.0/UDP <br>> > 4.68.250.148:5060;branch=z9hG4bK506071629460-1246361886000<br>> > From:"HIX <br>> > INC"<sip:+18592192438@4.68.250.148;isup-oli=0>;tag=VPSF506071629460<br>> > To:<sip:+19194393536@4.79.212.229:5060><br>> > Call-ID: ATLMGC0120090805185238005215@209.244.63.45<br>> > CSeq: 1 INVITE<br>> > Contact:<sip:+18592192438@4.68.250.148:5060;transport=udp><br>> > Max-Forwards: 68<br>> > Content-Type: application/sdp<br>> > Content-Length: 173<br>> ><br>> > v=0<br>> > o=- 1249498358 1249498359 IN IP4 63.215.29.149<br>> > s=-<br>> > c=IN IP4 63.215.29.149<br>> > t=0 0<br>> > m=audio 61030 RTP/AVP 0 18 101<br>> > a=rtpmap:101 telephone-event/8000<br>> > a=fmtp:101 0-15<br>> ><br>> ><br>> > ------------------------------------------------------------------------<br>> > Express your personality in color! Preview and select themes for <br>> > HotmailR. Try it now. <br>> ><br>> <http://www.windowslive-hotmail.com/LearnMore/personalize.aspx?ocid=PID23391<br>> ::T:WLMTAGL:ON:WL:en-US:WM_HYGN_express:082009> <br>> ><br>> > ------------------------------------------------------------------------<br>> ><br>> > _______________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> ><br>> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> > Register Now: http://www.astricon.net<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> <br>> -- <br>> Ishfaq Malik<br>> Software Developer<br>> PackNet Ltd<br>> <br>> Office: 0161 660 3062<br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> Register Now: http://www.astricon.net<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> <br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>> Register Now: http://www.astricon.net<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br /><hr />Get back to school stuff for them and cashback for you. <a href='http://www.bing.com/cashback?form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_BackToSchool_Cashback_BTSCashback_1x1' target='_new'>Try Bing now.</a></body>
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