<div dir="ltr">I think it might just be having the SDP passthrough on at all on a SIP to non-SIP call will do this. You could try some of the more intensive CCSIP debugs to see if there's a certain line it has a problem with but I think it's just the fact that it knows there will be SDP at one point and it won't be able to pass it through to the other leg due to non-SIP.</div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Oct 5, 2015 at 11:31 PM, Ryan Huff <span dir="ltr"><<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><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>Date: Mon, 5 Oct 2015 23:14:07 -0400<br>Subject: Re: [cisco-voip] Odd call behavior CME,CUBE,CUCM<br>From: <a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a><br>To: <a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a><br>CC: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><div><div class="h5"><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><br><div>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 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>
<br>_______________________________________________<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>
<br></blockquote></div><br></div></div></div></div> </div></div>
</blockquote></div><br></div>