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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'>Call Manager bug<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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal>CSCtb01167<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><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><b><span style='font-size:8.0pt;font-family:"Arial","sans-serif"'>CUCM fails to ACK the 200 OK after blind transfer over SIP trunk </span></b><span style='font-size:8.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></span></p></td></tr><tr><td valign=top style='padding:0in 6.0pt 6.0pt 6.0pt'><p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:8.0pt;font-family:"Arial","sans-serif"'>Symptom:</span></b><span style='font-size:8.0pt;font-family:"Arial","sans-serif"'><br><br>After blind transferring a call the transfer destination gets one-way voice<br>and eventually the call disconnects<o:p></o:p></span></p></td><td style='padding:3.75pt 3.75pt 3.75pt 3.75pt'></td></tr></table><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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><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><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"'> Mike Lydick [mailto:mike.lydick@gmail.com] <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<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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='margin-bottom:12.0pt'><br clear=all>Best Regards,<br><br>Mike Lydick<br><br><br><br><o:p></o:p></p><div><p class=MsoNormal>On Fri, Dec 17, 2010 at 11:24 AM, Bill Riley <<a href="mailto:bill@hitechconnection.net">bill@hitechconnection.net</a>> wrote:<o:p></o:p></p><p class=MsoNormal>main up during the transition to SRST? I do have MTP required o<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>