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well the good call is from bandwidth.com as example. we haven't had a good call form your office. they all fail. so i tried calling the same external number from each extension the a different external number from all three extension. they all fail. the guy at bandwidth.com just sent us that as a sample of what it should look like.<br><br><hr id="stopSpelling">From: danny@debsinc.com<br>To: asterisk-users@lists.digium.com<br>Date: Wed, 12 Aug 2009 11:42:07 -0500<br>Subject: Re: [asterisk-users] call drops after a few seconds<br><br>







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<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">So a “good” call works on all 3 lines and
a “bad” call fails on all 3?&nbsp; Are there numbers that alternate between good and
bad?</span></font></p>

<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

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<p class="EC_MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; 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
11:39 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></p>

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<p class="EC_MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">&nbsp;</span></font></p>

<p class="EC_MsoNormal" style="margin-bottom: 12pt;"><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;">yup just did all the same results</span></font></p>

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<p class="EC_MsoNormal" style="margin-bottom: 12pt;"><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;">From: danny@debsinc.com<br>
To: asterisk-users@lists.digium.com<br>
Date: Wed, 12 Aug 2009 11:14:43 -0500<br>
Subject: Re: [asterisk-users] call drops after a few seconds</span></font></p>

<div>

<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">So you have executed this call
scenario:&nbsp; 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?&nbsp; And then replicated on the “good” call (1-a,2-a…)?</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>

<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>

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<p class="EC_MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; 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><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>

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<p class="EC_MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">&nbsp;</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>

<p class="EC_MsoNormal" style="margin-bottom: 12pt;"><font size="2" face="Verdana"><span style="font-size: 10pt; 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>
&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>
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------------------------------------------------------------------------<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</span></font></p>

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