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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I tested this the other week with 8.0.3a and IOS 15.1.2T with a PRI and SIP Trunk back to CallManager. I recall with MTP Required checked (always a bad idea) and a local IOS Software MTP begin the only registered MTP in MRG/MRGL the call would not stay up upon invoking SRST by simulating wan failure. A good design should work to avoid/mitiigate MTP/Transcoding (assuming bandwidth is inexpensive).<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Justin Steinberg<br><b>Sent:</b> Monday, December 20, 2010 6:03 PM<br><b>To:</b> Bill Riley<br><b>Cc:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip] Call Preserve SIP<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>I'm not sure whether a MTP on the local device providing SRST would keep the call up. I tend to think it won't but that's just my opinion. If it was me dealing with this, I would upgrade CM to a version that has a fix for that defect and then remove the MTP from the call path.<o:p></o:p></p><div><p class=MsoNormal>On Mon, Dec 20, 2010 at 8:05 AM, Bill Riley <<a href="mailto:bill@hitechconnection.net">bill@hitechconnection.net</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'>Anyone else? Anyway to keep the call up when switching to SRST mode? </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:-moz-use-text-color -moz-use-text-color'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Bill Riley<br><b>Sent:</b> Friday, December 17, 2010 3:16 PM<br><b>To:</b> 'Buchanan, James'<o:p></o:p></span></p><div><div><p class=MsoNormal><span style='font-size:10.0pt'><br><b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] Call Preserve SIP<o:p></o:p></span></p></div></div></div></div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'>Yes same bug different issue. The bug causes me to need to use an MTP for transfers to work correctly even though I shouldn’t need one. My current problem is that when the WAN drops I lose my call to the SIP provider so I wonder if it is because I am using the software MTP on Call manager. </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:-moz-use-text-color -moz-use-text-color'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> Buchanan, James [mailto:<a href="mailto:jbuchanan@presidio.com" target="_blank">jbuchanan@presidio.com</a>] <br><b>Sent:</b> Friday, December 17, 2010 2:56 PM<br><b>To:</b> Bill Riley<br><b>Subject:</b> RE: [cisco-voip] Call Preserve SIP</span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'>Isn’t that the one you hit a while back?</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:7.0pt;color:#7F7F7F'>James Buchanan </span></b><span style='font-size:7.0pt;color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;color:#7F7F7F'> Technology Manager, UC </span></b><span style='font-size:7.0pt;color:#7E7E7E'>|</span><span style='font-size:7.0pt;color:#7F7F7F'> <b>South Region </b></span><span style='font-size:7.0pt;color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;color:#7F7F7F'> Presidio Networked Solutions <br>12 Cadillac Dr, Suite 130, Brentwood, TN 37027 </span></b><span style='font-size:7.0pt;color:#7E7E7E'>|</span><b><span style='font-size:7.0pt;color:#7F7F7F'> </span></b><b><span style='font-size:7.0pt;color:#7E7E7E'><a href="mailto:jbuchanan@ctiusa.com" target="_blank">jbuchanan@presidio.com</a><br></span></b><b><span style='font-size:7.0pt;color:#7F7F7F'>D: 615-866-5729</span></b><span style='font-size:7.0pt;color:#7F7F7F'> </span><span style='font-size:7.0pt;color:#7E7E7E'>| <b>F:</b> </span><b><span style='font-size:7.0pt;color:#7F7F7F'>615-866-5781</span></b><span style='font-size:7.0pt;color:#7F7F7F'> | </span><b><span style='font-size:7.0pt;color:#7E7E7E'><a href="http://www.presidio.com/" target="_blank">www.presidio.com</a></span></b><span style='font-size:7.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><b><span style='font-size:7.0pt;color:#7F7F7F'>CCIE #25863, Voice</span></b><o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;border-color:-moz-use-text-color -moz-use-text-color'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Bill Riley<br><b>Sent:</b> Friday, December 17, 2010 2:41 PM<br><b>To:</b> 'Mike Lydick'<br><b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] Call Preserve SIP</span><o:p></o:p></p></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'>Call Manager bug</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>CSCtb01167<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><table class=MsoNormalTable border=0 cellpadding=0 width="100%" style='width:100.0%'><tr><td colspan=2 style='padding:6.0pt 6.0pt 6.0pt 6.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:8.0pt'>CUCM fails to ACK the 200 OK after blind transfer over SIP trunk </span></b><o:p></o:p></p></td></tr><tr><td valign=top style='padding:0in 6.0pt 6.0pt 6.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><b><span style='font-size:8.0pt'>Symptom:</span></b><span style='font-size:8.0pt'><br><br>After blind transferring a call the transfer destination gets one-way voice<br>and eventually the call disconnects</span><o:p></o:p></p></td><td style='padding:3.75pt 3.75pt 3.75pt 3.75pt'></td></tr></table><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> Mike Lydick [mailto:<a href="mailto:mike.lydick@gmail.com" target="_blank">mike.lydick@gmail.com</a>] <br><b>Sent:</b> Friday, December 17, 2010 11:12 AM<br><b>To:</b> Bill Riley<br><b>Subject:</b> Re: [cisco-voip] Call Preserve SIP</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>What is requiring the MTP? Is there an early offer or DTMF codec/media codec mismatch? <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><br clear=all>Best Regards,<br><br>Mike Lydick<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>On Fri, Dec 17, 2010 at 11:24 AM, Bill Riley <<a href="mailto:bill@hitechconnection.net" target="_blank">bill@hitechconnection.net</a>> wrote:<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>main up during the transition to SRST? I do have MTP required o<o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>
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