<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<div><br><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><div><br></div><div>!!!!!!!!!!!!!!!!!!!<br><div><div> </div><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><div><br></div><div>here is where i think i should change to read like so : "midcall-signalling black" instead of passthru<br></div><div><br></div><div><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></div><div><br></div><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><div><br></div><div>ISP -----Their SIPtrunk----MyCube----CUCM10.<wbr>5----VoiceVlan-----7975, and 8861's</div><div><br></div><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><div><br></div><div>please help<br clear="all"><div><br></div><div class="gmail-m_7797918839443430891gmail_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 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>
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