I have been.<br><br>I've made several changes at once & now inbound calls calls stay up longer than 29:45.<br><br>25.30 is cube.<br><br>25.21 is subscriber CCM 7.1.3.20000-2.<br><br>Dec 30 11:53:57.974 EST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>BYE sip:8340636@192.168.25.30:5060;transport=tcp SIP/2.0<br><br>Reason: Q.850;cause=41<br><br>Date: Wed, 30 Dec 2009 16:24:21 GMT<br><br>From: "Edens and Avant" <<a href="mailto:sip%3A8037442493@192.168.25.21">sip:8037442493@192.168.25.21</a>>;tag=4487f683-0ba3-49ae-ae40-51c85814ef6d-46970603<br>
<br>Content-Length: 0<br><br>User-Agent: Cisco-CUCM7.1<br><br>To: <<a href="mailto:sip%3A8340636@192.168.25.30">sip:8340636@192.168.25.30</a>>;tag=F83BA30-49E<br><br>Call-ID: <a href="mailto:c7beae00-b3b17eb5-11554-1519a8c0@192.168.25.21">c7beae00-b3b17eb5-11554-1519a8c0@192.168.25.21</a><br>
<br>Via: SIP/2.0/TCP 192.168.25.21:5060;branch=z9hG4bK120b2658de523<br><br>CSeq: 102 BYE<br><br>Max-Forwards: 70<br><br><div class="gmail_quote">On Wed, Dec 30, 2009 at 12:35 PM, Adam Frankel <span dir="ltr"><<a href="mailto:afrankel@cisco.com">afrankel@cisco.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="font-size: small; font-family: Arial;" bgcolor="#ffffff" text="#000000">
<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><font color="#888888">
<pre cols="72">Adam </pre>
<br>
</font><span style="color: rgb(0, 0, 0);"><div class="im"><br>
<br>
------------Original Message--------------<br>
From: Lee <a href="mailto:ender9600@gmail.com" target="_blank"><ender9600@gmail.com></a><br>
Sent: Wed, Dec 30, 2009 8:06:03 Am<br>
To: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
CC: <br></div><div><div></div><div class="h5">
Subject: [cisco-voip] sip call disconnects every 29min 45 sec</div></div></span>
<blockquote style="border: medium none ! important; padding-left: 0px ! important; padding-right: 0px ! important; margin-left: 0px ! important; margin-right: 0px ! important;" type="cite"><div><div></div><div class="h5">
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>
</div></div><pre><div class="im"><hr size="4" width="90%">
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</blockquote>
</div>
</blockquote></div><br>