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Have a read through this thread: <a href="https://supportforums.cisco.com/t5/ip-telephony/issues-with-cucm-sip-trunk-and-hold/td-p/2395620">https://supportforums.cisco.com/t5/ip-telephony/issues-with-cucm-sip-trunk-and-hold/td-p/2395620</a><br>
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<div id="AppleMailSignature">Sent from my iPhone</div>
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On Feb 22, 2018, at 13:23, Jonatan Quezada <<a href="mailto:jonatan.quezada@chemeketa.edu">jonatan.quezada@chemeketa.edu</a>> wrote:<br>
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<div dir="ltr">and if so, do I need to modify this portiopn of my cube configuration?, This guy seems to describe what I am seeing on our main campus
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<div><a href="https://supportforums.cisco.com/t5/video-over-ip/sip-trunk-call-hold-fails-no-audio-no-resume-no-sdp-from-cube/td-p/2214595" target="_blank">https://supportforums.cisco.<wbr>com/t5/video-over-ip/sip-<wbr>trunk-call-hold-fails-no-<wbr>audio-no-resume-no-sdp-from-<wbr>cube/td-p/2214595</a></div>
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<div>!!!!!!!!!!!!!!!!!!!<br>
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<div>voice service voip</div>
<div> address-hiding</div>
<div> dtmf-interworking rtp-nte</div>
<div> mode border-element license capacity 500</div>
<div> media bulk-stats</div>
<div> allow-connections sip to sip</div>
<div> no supplementary-service sip moved-temporarily</div>
<div> redirect ip2ip</div>
<div> fax protocol t38 nse version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw</div>
<div> sip</div>
<div> rel1xx supported "rel100"</div>
<div> session refresh</div>
<div> asserted-id pai</div>
<div> privacy pstn</div>
<div> localhost dns:<a href="http://voip.centurylink.com" target="_blank">voip.centurylink.com</a></div>
<div> no update-callerid</div>
<div> early-offer forced</div>
<div><font size="4"> midcall-signaling passthru</font></div>
<div><font size="4">......</font></div>
<div>begin my notes:</div>
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<div>here is where i think i should change to read like so : "midcall-signalling black" instead of passthru<br>
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<div>4431-voice-gw(conf-serv-sip)#mid</div>
<div>4431-voice-gw(conf-serv-sip)#midcall-signaling ?</div>
<div> block Block all SIP messages in midcall</div>
<div> passthru Passthrough SIP messages from one IP leg to another IP leg</div>
<div> preserve-codec preserve initial negotiated codec i.e. midcall codec change</div>
<div> denial</div>
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<div>.....</div>
<div> privacy-policy passthru</div>
<div> pass-thru subscribe-notify-events all</div>
<div> pass-thru content sdp</div>
<div> sip-profiles 100</div>
<div> no call service stop</div>
<div>!</div>
<div>here is my call flow</div>
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<div>ISP -----Their SIPtrunk----MyCube----CUCM10.<wbr>5----VoiceVlan-----7975, and 8861's</div>
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<div>the phones here and there are experiencing, one way audio, and transfer drops, not all of the time though. what do we suggest?</div>
<div>why would you want to block SDP updates mid call, how will CUCM know how to changes call characteristics?</div>
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<div>please help<br clear="all">
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<div dir="ltr"><font face="arial, helvetica, sans-serif" size="4"><br>
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<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>
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<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>
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<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>
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