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