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<font size="-1"><font face="Arial">Probably good to start with the
debug ccsip to see who is dropping the call and why.</font></font><br>
<pre class="moz-signature" cols="72">Adam </pre>
<br>
<span style="color: rgb(0, 0, 0);" class="headerSpan"><br>
<br>
------------Original Message--------------<br>
From: Lee <a class="moz-txt-link-rfc2396E" href="mailto:ender9600@gmail.com"><ender9600@gmail.com></a><br>
Sent: Wed, Dec 30, 2009 8:06:03 Am<br>
To: <a class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
CC: <br>
Subject: [cisco-voip] sip call disconnects every 29min 45 sec</span>
<blockquote
style="border: medium none ! important; padding-left: 0px ! important; padding-right: 0px ! important; margin-left: 0px ! important; margin-right: 0px ! important;"
cite="mid:cef0c4560912300506n353c96b9hb6cba6198a1c848@mail.gmail.com"
type="cite">SIP Provider is Paetec. CCM7 to 2821 Router running Cube
to Provider. Using Trunk from CCM7 to Cube.<br>
<br>
voice service voip <br>
allow-connections sip to sip<br>
no supplementary-service sip moved-temporarily<br>
no fax-relay sg3-to-g3 <br>
h323<br>
sip<br>
header-passing error-passthru<br>
early-offer forced<br>
midcall-signaling passthru<br>
min-se 1700<br>
!<br>
!<br>
voice class codec 1<br>
codec preference 1 g711ulaw<br>
codec preference 2 g729r8<br>
<br>
dial-peer voice 1000 voip<br>
destination-pattern 1[2-9]..[2-9]......<br>
voice-class codec 1<br>
no voice-class sip localhost <br>
no voice-class sip outbound-proxy <br>
voice-class sip early-offer forced<br>
session protocol sipv2<br>
session target sip-server<br>
incoming called-number ....<br>
dtmf-relay rtp-nte<br>
ip qos dscp cs3 signaling<br>
<br>
dial-peer voice 8031 voip<br>
preference 3<br>
destination-pattern 803.......<br>
voice-class codec 1<br>
no voice-class sip outbound-proxy <br>
session protocol sipv2<br>
session target ipv4:192.168.xx.21<br>
dtmf-relay rtp-nte<br>
<br>
sip-ua <br>
credentials username xxxxxxxxxx password 7 xxx445415F realm none<br>
authentication username xxxxxxxxxx password 7 xxx756085F<br>
no remote-party-id<br>
retry invite 2<br>
retry register 10<br>
timers connect 100<br>
registrar ipv4:172.31.255.37 expires 3600<br>
sip-server ipv4:172.31.255.37<br>
presence enable<br>
<br>
I've done test calls during the day & after hours - all calls drop
after 29 min 45 sec. Any suggestions?<br>
<br>
Lee<br>
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</pre>
</blockquote>
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