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<body class='hmmessage'><div dir='ltr'>HA! Thanks Brian ... orphaned passthru .... this was the 28th (seriously) time I reviewed the config ... finally spotted it.<br><br>Is this because CUBE is presenting DTMF as part of the INVITE?<br><br>Thanks,<br><br>Ryan<br><br><div><hr id="stopSpelling">Date: Mon, 5 Oct 2015 23:14:07 -0400<br>Subject: Re: [cisco-voip] Odd call behavior CME,CUBE,CUCM<br>From: bmeade90@vt.edu<br>To: ryanhuff@outlook.com<br>CC: cisco-voip@puck.nether.net<br><br><div dir="ltr">Make sure you don't have any of these on the dial-peer:<div><span style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);">voice-class sip pass-thru headers unsupp</span><br style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);"><span style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);">voice-class sip pass-thru content unsupp</span><br style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);"><span style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);">voice-class sip pass-thru content sdp</span><br></div><div><span style="color:rgb(0,0,0);font-family:'Helvetica Neue',Helvetica,Arial,sans-serif;font-size:14.4px;line-height:21.6px;white-space:pre-wrap;background-color:rgb(240,240,240);"><br></span></div><div><span style="background-color:rgb(240,240,240);"><font color="#000000" face="Helvetica Neue, Helvetica, Arial, sans-serif"><span style="font-size:14.4px;line-height:21.6px;white-space:pre-wrap;"><a href="http://www.dslreports.com/forum/r27144625-Config-CME-8-Help-with-Incoming-calls-on-SIP" target="_blank">http://www.dslreports.com/forum/r27144625-Config-CME-8-Help-with-Incoming-calls-on-SIP</a></span></font><br></span></div></div><div class="ecxgmail_extra"><br><div class="ecxgmail_quote">On Mon, Oct 5, 2015 at 11:00 PM, Ryan Huff <span dir="ltr"><<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>></span> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">
<div><div dir="ltr">I have CME that has a SIP phone registered to it and a SCCP phone registered to it.<br>SIP=4002<br>SCCP=4001<br> <br>I have a call path that looks like:<br> <br>CUCM->SIPTrunk->CUBE->SIPTrunk->CME<br> <br>When I dial the SIP phone (on CME) from call manager the call completes and goes fine. When I dial the skinny phone (on CME) from call manager the call immediately fails. However, from CME, using the skinny phone, I can dial anything in call manager and the call completes.<br> <br>CCSIP on CME (for one of the failed calls) shows the INVITE then TRYING then SIP/2.0 500 Internal Server Error. CCAPI INOUT shows a cause code of 16, but this was anything but a normal clearing. Just for the hell of it, I also added a transcoder in CME and registered it in telephony-service, just to see if there was a codec flying around that I wasn't seeing .... but alas, no.<br> <br>I've attached the ccsip and ccapi inout; I think I have just been starring at it too long ... I can't see it.<br><br>Thanks,<br><br>Ryan<br> <br> </div></div>
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