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.<br>
<br><div class="gmail_quote">On Mon, Dec 20, 2010 at 8:05 AM, Bill Riley <span dir="ltr"><<a href="mailto:bill@hitechconnection.net">bill@hitechconnection.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Anyone else? Anyway to keep the call up when switching to SRST mode? </span></p><p class="MsoNormal">
<span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><div><div style="border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <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'<div><div></div><div class="h5"><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</div></div></span></p></div></div><div><div></div><div class="h5"><p class="MsoNormal"> </p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">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></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><div><div style="border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> 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></p></div></div><p class="MsoNormal"> </p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Isn’t that the one you hit a while back?</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><div><p class="MsoNormal"><b><span style="font-size: 7pt; color: rgb(127, 127, 127);">James Buchanan </span></b><span style="font-size: 7pt; color: rgb(126, 126, 126);">|</span><b><span style="font-size: 7pt; color: rgb(127, 127, 127);"> Technology Manager, UC </span></b><span style="font-size: 7pt; color: rgb(126, 126, 126);">|</span><span style="font-size: 7pt; color: rgb(127, 127, 127);"> <b>South Region </b></span><span style="font-size: 7pt; color: rgb(126, 126, 126);">|</span><b><span style="font-size: 7pt; color: rgb(127, 127, 127);"> Presidio Networked Solutions <br>
12 Cadillac Dr, Suite 130, Brentwood, TN 37027 </span></b><span style="font-size: 7pt; color: rgb(126, 126, 126);">|</span><b><span style="font-size: 7pt; color: rgb(127, 127, 127);"> </span></b><b><span style="font-size: 7pt; color: rgb(126, 126, 126);"><a href="mailto:jbuchanan@ctiusa.com" target="_blank">jbuchanan@presidio.com</a><br>
</span></b><b><span style="font-size: 7pt; color: rgb(127, 127, 127);">D: 615-866-5729</span></b><span style="font-size: 7pt; color: rgb(127, 127, 127);"> </span><span style="font-size: 7pt; color: rgb(126, 126, 126);">| <b>F:</b> </span><b><span style="font-size: 7pt; color: rgb(127, 127, 127);">615-866-5781</span></b><span style="font-size: 7pt; color: rgb(127, 127, 127);"> | </span><b><span style="font-size: 7pt; color: rgb(126, 126, 126);"><a href="http://www.presidio.com/" target="_blank">www.presidio.com</a></span></b><span style="font-size: 7pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal" style="margin-bottom: 12pt;"><b><span style="font-size: 7pt; color: rgb(127, 127, 127);">CCIE #25863, Voice</span></b><span style="font-size: 11pt; color: rgb(31, 73, 125);"></span></p></div><p class="MsoNormal">
<span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><div><div style="border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <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></p>
</div></div><p class="MsoNormal"> </p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Call Manager bug</span></p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal">CSCtb01167</p><p class="MsoNormal"> </p><p class="MsoNormal"> </p><table style="width: 100%;" width="100%" border="0" cellpadding="0">
<tbody><tr><td colspan="2" style="padding: 6pt;"><p class="MsoNormal"><b><span style="font-size: 8pt;">CUCM fails to ACK the 200 OK after blind transfer over SIP trunk </span></b><span style="font-size: 8pt;"></span></p></td>
</tr><tr><td style="padding: 0in 6pt 6pt;" valign="top"><p class="MsoNormal" style="margin-bottom: 12pt;"><b><span style="font-size: 8pt;">Symptom:</span></b><span style="font-size: 8pt;"><br><br>After blind transferring a call the transfer destination gets one-way voice<br>
and eventually the call disconnects</span></p></td><td style="padding: 3.75pt;"></td></tr></tbody></table><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> 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></p><p class="MsoNormal"> </p><p class="MsoNormal">What is requiring the MTP? Is there an early offer or DTMF codec/media codec mismatch? </p>
<div><p class="MsoNormal" style="margin-bottom: 12pt;"><br clear="all">Best Regards,<br><br>Mike Lydick</p><div><p class="MsoNormal">On Fri, Dec 17, 2010 at 11:24 AM, Bill Riley <<a href="mailto:bill@hitechconnection.net" target="_blank">bill@hitechconnection.net</a>> wrote:</p>
<p class="MsoNormal">main up during the transition to SRST? I do have MTP required o</p></div><p class="MsoNormal"> </p></div></div></div></div></div><br>_______________________________________________<br>
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