<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
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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'>CallManager has a Service Parameter something like stream full duplex, it has fixed similar issues for me before.<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'><a href="http://www.gossamer-threads.com/lists/cisco/voip/147274?page=last">http://www.gossamer-threads.com/lists/cisco/voip/147274?page=last</a><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><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><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>Mark Holloway<br><b>Sent:</b> Wednesday, August 29, 2012 8:42 AM<br><b>To:</b> Divin John<br><b>Cc:</b> Cisco VoIPoE List<br><b>Subject:</b> Re: [cisco-voip] CUCM SIP Trunk and Call Hold (VZB, no audio)<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><br><br>Are there any docs or links where this issue is discussed? Thanks.<br><br>On Aug 27, 2012, at 3:29 AM, Divin John wrote:<br><br>> It could be the classic a=inactive problem when on hold.<br>> <br>> Get a packet capture or a trace from CUCM and track the SIP leg.<br>> <br>> On 25/08/12 2:09 AM, "Mark Holloway" <<a href="mailto:mh@markholloway.com">mh@markholloway.com</a>> wrote: <br>> <br>>> Anyone experience an issue with a Verizon SIP Trunk where you put a PSTN<br>>> call on hold, then take it off hold and have no audio? This issue<br>>> happened on both CUCM 6.1 and 8.6, so maybe it's the way the customer<br>>> configured it.<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> <br>>> <br>> <br>> <br><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> <br><br><br><span style='color:white'>itevomcid</span> <o:p></o:p></p></div></body></html>