<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=utf-8">
</head>
<body dir="auto">
<div><span></span></div>
<div>
<div><span></span></div>
<div>
<div><span></span></div>
<div>Empirically, this “looks” like one way audio. How long will the call stay connected? Indefinitely? 30 seconds? 2 minutes?</div>
<div><br>
</div>
<div>Your call doesn’t appear to have a need for MTP or Transcoding (G711 both sides and matching sample sizes); so I wouldn’t start there.</div>
<div><br>
</div>
<div>Check these items and see what you find;</div>
<div><br>
</div>
<div>CUBE: ip trusted address list (make sure all provider signaling and media addresses are authorized or ip authentication is off (which I do not recommend) and make sure you include any CUCM addresses that are not used in dial peers).</div>
<div><br>
</div>
<div>CUBE: double check your media and signal bindings and make sure they are binding correctly. Are you globally binding or dial peer binding?</div>
<div><br>
</div>
<div>CUCM: verify the SIP trunk points to the CUBE interface that signaling is bound to (generally the same interface media would be bound to as well).</div>
<div><br>
</div>
<div>CUBE: </div>
<div>#logging buffered 10000000</div>
<div>#enable debug ccsip messages</div>
<div><br>
</div>
<div>Place a call and then look at the logs. Do you see any SIP error messages in the 4xx, 5xx (or more rare 6xx) range?</div>
<div><br>
</div>
<div>As a quick gut check, if you can, enable “MTP Required” on the CUCM SIP trunk facing the CUBE (and make sure it has access to an MRGL/MRG that uses a CUCM node for MTP) and reset the trunk and test a call. If this works, it likely means you’re facing a
network path issue between the phone’s IP network and the network of the CUBE interface facing CUCM.</div>
<div><br>
</div>
<div>Outside of that, like Anthony said, it could be almost anything. A “sh run” or “sh tech” on the cube with a logging buffer from a ccsip messages during a failed call will generally get the ball rolling for most of us on this list in terms of offering targeted
assistance.</div>
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Ryan</div>
<div>
<div><br>
On Feb 5, 2018, at 2:37 PM, 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">The fact that you received 2 packets is interesting. Tells me that there is routing happening correctly...to some degree.
<div><br>
</div>
<div>If you go to the web page of the phone and click on stream 1, does the far end IP address match your CUBE address?</div>
<div><br>
</div>
<div>Also, there's a lot of settings that need to be considered when implementing SIP, such as:</div>
<div><br>
</div>
<div>Early Offer and MTP usage</div>
<div>PRACK/Early Media</div>
<div>Offfer/Answer (Capabilities)</div>
<div>Interface Binding</div>
<div>Transport Protocol</div>
<div>OPTIONS Ping</div>
<div>Duplex Streaming</div>
<div>Midcall Signaling</div>
<div>Timers</div>
<div>etc.</div>
<div><br>
</div>
<div>Depending on your setting, a lot of different possibilities exist for why you might have the experience you have. If you could paint a clearer picture of your scenario, that might help out.</div>
<div><br>
<div class="gmail_quote">
<div dir="ltr">On Fri, Feb 2, 2018 at 5:47 PM Jonatan Quezada <<a href="mailto:jonatan.quezada@chemeketa.edu">jonatan.quezada@chemeketa.edu</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">I get that this is usually routing but, is it also routing when the issue is intermittent?
<div><br>
</div>
<div>our call flow is like so</div>
<div><br>
</div>
<div>CentLink(Provider) ----siptrunk30Meg-PPP(IQ-private)---Cube---CUCM10.5, uccx,unity</div>
<div><br>
</div>
<div><image.png><br>
</div>
<div><br>
</div>
<div>bonus facts, I have an operator who is in one of the two most affected buildings and she can recover the call after hold, resume,hold,resume sequence. then full rtp stream is there and she can hear and speak with caller.</div>
<div><br>
</div>
<div>are there SIP state change timers I can adjust, I want to tread lightly though because out of all of our outreachs seperated by a metro ethernet hub and spoke topology and almost 30 buildings here on main campus only 2 seem to be affected.</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br clear="all">
<div><br>
</div>
-- <br>
<div class="m_9197210602978245346gmail_signature">
<div dir="ltr">
<div>
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div><font face="arial, helvetica, sans-serif" size="2">For immediate assistance please reach out to Chemeketa IT Help Desk at
<a href="tel:(503)%20399-7899" value="+15033997899" target="_blank">5033997899</a></font></div>
<div><font face="arial, helvetica, sans-serif" size="2">-or-</font></div>
<div><font face="arial, helvetica, sans-serif" size="2">Visit the help center from your employee dashboard found here:</font></div>
<div><font face="arial, helvetica, sans-serif" size="4"><b><a href="https://dashboard.chemeketa.edu/helpcenter/default.aspx" target="_blank">https://dashboard.chemeketa.edu/helpcenter/default.aspx</a></b><br>
</font></div>
<div><font face="arial, helvetica, sans-serif" size="4"><br>
</font></div>
<div dir="ltr"><font face="arial, helvetica, sans-serif" size="4"><br>
</font></div>
<div dir="ltr"><font face="arial, helvetica, sans-serif" size="4">Johnny Q</font></div>
<div dir="ltr"><span style="font-family:arial,helvetica,sans-serif;font-size:small">Voice Technology Analyst - TelNet</span><font face="arial, helvetica, sans-serif" size="4"><br>
</font>
<div><font face="arial, helvetica, sans-serif" size="4">Chemeketa Community College</font></div>
<div><span style="font-size:12.8px"><a href="mailto:Johnny.Q@chemeketa.edu" target="_blank">Johnny.Q@chemeketa.edu</a></span></div>
<div><font size="1"><span style="font-family:arial,helvetica,sans-serif">Building 22 Room 131</span><br>
</font></div>
<div><font face="arial, helvetica, sans-serif" size="1">Work <a href="tel:(503)%20399-5294" value="+15033995294" target="_blank">
5033995294</a></font></div>
<div><font face="arial, helvetica, sans-serif" size="1">Mobile <a href="tel:(971)%20218-2110" value="+19712182110" target="_blank">
9712182110</a></font></div>
<div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">SIP <a href="tel:(503)%20540-6686" value="+15035406686" target="_blank">
5035406686</a></span><br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
_______________________________________________<br>
cisco-voip mailing list<br>
<a href="mailto:cisco-voip@puck.nether.net" target="_blank">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>
</blockquote>
</div>
</div>
</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>
</div>
</blockquote>
</div>
</div>
</div>
</body>
</html>