<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div><div><span style="background-color: rgba(255, 255, 255, 0);">Hi All,</span></div><div><span style="background-color: rgba(255, 255, 255, 0);"><br></span></div><div><span style="background-color: rgba(255, 255, 255, 0);">I think the config looks correct;</span></div><div><span style="background-color: rgba(255, 255, 255, 0);"><br></span></div><div><span style="background-color: rgba(255, 255, 255, 0);">- Dial-peer 1 is the dial-peer you want to record so you apply media-class 30</span></div><div><span style="background-color: rgba(255, 255, 255, 0);">- Media-class 30 is associated with recorder 400</span></div><div><span style="background-color: rgba(255, 255, 255, 0);">- Recorder 400 is associated with media-recording 3 (in other words dial-peer 3)</span></div><div><span style="background-color: rgba(255, 255, 255, 0);">- Dial-peer 3 is the 'SIP Trunk' towards MediaSense</span></div><div><span style="background-color: rgba(255, 255, 255, 0);"><br></span></div><div><span style="background-color: rgba(255, 255, 255, 0);">On MediaSense you would need to make sure 450123 is configured to record but I'm sure you've configured that already.</span></div><div><span style="background-color: rgba(255, 255, 255, 0);"><br></span></div><div><span style="background-color: rgba(255, 255, 255, 0);">I've had some really weird issues with MediaSense in the past where CUCM was sending TCP SYN on port 5060 but MediaSense never responded. A cluster reboot of MediaSense solved that issue. Perhaps take an IP Traffic Export on the router to see if it is sending TCP SYN and if MediaSense is responding.</span></div><div><span style="background-color: rgba(255, 255, 255, 0);"><br></span><div><span style="background-color: rgba(255, 255, 255, 0);">Sent from my iPhone</span></div></div></div><div><br>On 2 Apr 2016, at 02:02, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com">avholloway+cisco-voip@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">First of all, be careful doing this in production:<div><br></div><blockquote style="margin:0 0 0 40px;border:none;padding:0px"><div><div><font face="monospace, monospace">voice service voip</font></div></div><div><div><font face="monospace, monospace"> ip address trusted list</font></div></div><div><div><font face="monospace, monospace"> ipv4 0.0.0.0 0.0.0.0</font></div></div></blockquote><div><br></div><div>That is just reducing the security of your application and opening you up to abuse. It's fine for troubleshooting and eliminating it as root cause, but then remove it and add addresses/subnets in there to lock down from where you will accept control traffic from.</div><div><br></div><div>One last thing on this topic, since your dial-peers 2 and 3 already point to IP addresses of SIP peers, you don't need to even do anything more. That simple fact already permits those IP addresses to send you control traffic.</div><div><br></div><div>Ok, on to the recording bit. I have not done this task myself, but looking quickly through the following document:</div><div><br></div><div><a href="http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-ntwk-based.html">http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-ntwk-based.html</a><br></div><div><br></div><div>...it looks like you might have at least one error in your configuration.</div><div><br></div><div>The one error I think you have: Your "<font face="monospace, monospace"><b>media-class 30</b></font>" dial-peer command should be on dial-peer 3, not dial-peer 1.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Apr 1, 2016 at 3:56 AM, daniele visaggio <span dir="ltr"><<a href="mailto:visaggio.daniele@gmail.com" target="_blank">visaggio.daniele@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div>Good morning,<br><br></div>I'm trying to record calls via CUBE. It doesn't work. This means that on the recording server I can't see any SIP invite incoming from CUBE.<br><br></div>Scenario:<br><br></div>Phone --- CUCM --- SIP --- CUBE ---- ITSP ---- PSTN<br> |<br> |<br></div> Recording Server<br><br><br></div>Let's say I want to record all calls going to the PSTN.<br><br></div>This is my config:<br><br>#####<br><font size="2"><span style="font-family:arial,helvetica,sans-serif">!<br>voice service voip<br> ip address trusted list<br> ipv4 0.0.0.0 0.0.0.0<br> allow-connections sip to sip<br></span></font><span lang="en-US"><div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">!</span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">media profile recorder 400</span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">media-recording 3</span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>!</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>media class 30</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>recorder profile 400</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>!</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">!</span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">dial-peer voice 1 voip<br></span></span></span></font></div><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">description :: Incoming calls from CUCM ::<br></span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span><span lang="it">session protocol sipv2</span></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>incoming called-number .</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>media-class 30</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>codec g711ulaw</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>!</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>dial-peer voice 2 voip<br></span></span></font></div><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>description :: To ITSP/PSTN ::<br></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>destination-pattern 0T</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>session protocol sipv2</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>session target ipv4:10.128.179.12</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>codec g711ulaw</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>!</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>dial-peer voice 3 voip<br></span></span></font></div><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>description :: To Recorder Server ::<br></span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>destination-pattern 450123</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>session protocol sipv2</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>session target ipv4:10.130.221.218</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>codec g711ulaw</span></span></font></div><font size="2"><span style="font-family:arial,helvetica,sans-serif">
</span></font><div style="margin:0px"><font size="2"><span style="font-family:arial,helvetica,sans-serif"><span>!</span></span></font></div><span style="font-family:arial,helvetica,sans-serif">
</span></div><span style="font-family:arial,helvetica,sans-serif">
</span></span><br><div><br></div><div>I double checked the configuration and it seems correct to me. <br><br></div><div>Is there something else I need to do? Can someone spot an error? <br></div><div><div><div><div><div><div><br><br></div><div>Thank you,<br><br></div><div>Daniele<br></div><div><br></div></div></div></div></div></div></div>
<br>_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" rel="noreferrer" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
<br></blockquote></div><br></div>
</div></blockquote><blockquote type="cite"><div><span>_______________________________________________</span><br><span>cisco-voip mailing list</span><br><span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a></span><br><span><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a></span><br><span></span><br></div></blockquote></body></html>