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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>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…)?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Ott Rose<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 12, 2009
11:08 AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] call
drops after a few seconds</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 face=Verdana><span
style='font-size:10.0pt;font-family:Verdana'>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: <st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial
Discussion</st1:PersonName><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>
> >
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>
<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 <st1:place w:st="on"><st1:City w:st="on">Phoenix</st1:City>,
<st1:State w:st="on">Arizona</st1:State></st1:place><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 <st1:place w:st="on"><st1:City w:st="on">Phoenix</st1:City>,
<st1:State w:st="on">Arizona</st1:State></st1:place><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 <st1:place w:st="on"><st1:City w:st="on">Phoenix</st1:City>,
<st1:State w:st="on">Arizona</st1:State></st1:place><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<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Verdana><span style='font-size:10.0pt;
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