<html><head><base href="x-msg://26/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Thanks, Jason. I also saw the following in the CUCM system guide. I think the customer will sanity check this first, but if it doesn't fix the issue then I'll have them explore the full duplex stream.<div><br></div><div><br></div><div><span class="Apple-style-span" style="font-family: Calibri, sans-serif; font-size: 14px; "><div><span class="content"><table class="steptable" id="wp1150004table1149998" border="1" bordercolor="#808080" cellpadding="3" cellspacing="0" width="80%" style="position: static; z-index: auto; "><tbody><tr align="left" valign="top"><td><div><br class="webkit-block-placeholder"></div></td><td><a name="wp1162385"></a><p class="pB1_Body1"><span class="Apple-style-span" style="background-color: rgb(255, 255, 0); ">To prevent Cisco Unified Communications Manager from sending an INVITE a=inactive SDP message during call hold or media break during supplementary services, edit the appropriate SIP profile, and check the </span><span class="Apple-style-span" style="background-color: rgb(255, 255, 0); "><b class="cBold">Send send-receive SDP in mid-call INVITE</b></span><span class="Apple-style-span" style="background-color: rgb(255, 255, 0); "> check box.</span></p><span class="Apple-style-span" style="background-color: rgb(255, 255, 0); "><a name="wp1162861"></a></span><p class="pNT_NoteTable"><b>Note </b><img src="http://www.cisco.com/en/US/i/templates/blank.gif" alt="" border="0" height="2" width="1">This check box applies only to early offer enabled SIP trunks and has no impact on SIP line calls.</p><a name="wp1162713"></a><p class="pB1_Body1">When you enable <b class="cBold">Send send-receive SDP in mid-call INVITE </b>for an early offer SIP trunk in tandem mode, Cisco Unified Communications Manager inserts MTP to provide sendrecv SDP when a SIP device sends offer SDP with a=inactive or sendonly or recvonly in audio media line. In tandem mode, Cisco Unified Communications Manager depends on the SIP devices to initiate reestablishment of media path by sending either a delayed offer INVITE or mid-call INVITE with send-recv SDP.</p><a name="wp1162720"></a><p class="pB1_Body1">When you enable both <b class="cBold">Send send-receive SDP in mid-call INVITE</b> and <b class="cBold">Require SDP Inactive Exchange for Mid-Call Media Change</b> on the same SIP profile, the <b class="cBold">Send send-receive SDP in mid-call INVITE</b> setting overrides the <b class="cBold">Require SDP Inactive Exchange for Mid-Call Media Change</b> setting, so<b class="cBold"></b>Cisco Unified Communications Manager does not send an INVITE with a=inactive SDP in mid-call codec updates. For SIP line side calls, the <b class="cBold">Require SDP Inactive Exchange for Mid-Call Media Change</b> check box applies when enabled.</p><a name="wp1162788"></a><p class="pNT_NoteTable"><b>Note </b><img src="http://www.cisco.com/en/US/i/templates/blank.gif" alt="" border="0" height="2" width="1">To prevent the SDP mode from being set to inactive in a multiple-hold scenario, set the Duplex Streaming Enabled clusterwide service parameter (<b class="cBold">System > Service Parameters</b>) to <em class="cEmphasis">True</em>. </p></td></tr></tbody></table></span></div><div><div></div></div></span><div><div>On Aug 29, 2012, at 9:55 AM, Jason Aarons (AM) wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div lang="EN-US" link="blue" vlink="purple"><div class="WordSection1" style="page: WordSection1; "><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">CallManager has a Service Parameter something like stream full duplex, it has fixed similar issues for me before.<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><a href="http://www.gossamer-threads.com/lists/cisco/voip/147274?page=last" style="color: blue; text-decoration: underline; ">http://www.gossamer-threads.com/lists/cisco/voip/147274?page=last</a><o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><span style="font-size: 11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: 0in; "><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip-bounces@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip-bounces@puck.nether.net</a><span class="Apple-converted-space"> </span>[mailto:cisco-voip-bounces@puck.nether.net]<span class="Apple-converted-space"> </span><b>On Behalf Of<span class="Apple-converted-space"> </span></b>Mark Holloway<br><b>Sent:</b><span class="Apple-converted-space"> </span>Wednesday, August 29, 2012 8:42 AM<br><b>To:</b><span class="Apple-converted-space"> </span>Divin John<br><b>Cc:</b><span class="Apple-converted-space"> </span>Cisco VoIPoE List<br><b>Subject:</b><span class="Apple-converted-space"> </span>Re: [cisco-voip] CUCM SIP Trunk and Call Hold (VZB, no audio)<o:p></o:p></span></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "><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>><span class="Apple-converted-space"> </span><br>> Get a packet capture or a trace from CUCM and track the SIP leg.<br>><span class="Apple-converted-space"> </span><br>> On 25/08/12 2:09 AM, "Mark Holloway" <<a href="mailto:mh@markholloway.com" style="color: blue; text-decoration: underline; ">mh@markholloway.com</a>> wrote:<span class="Apple-converted-space"> </span><br>><span class="Apple-converted-space"> </span><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>>><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><span class="Apple-converted-space"> </span><br>>><span class="Apple-converted-space"> </span><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><span class="Apple-converted-space"> </span><br>>><span class="Apple-converted-space"> </span><br>><span class="Apple-converted-space"> </span><br>><span class="Apple-converted-space"> </span><br><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" style="color: blue; text-decoration: underline; ">cisco-voip@puck.nether.net</a><span class="Apple-converted-space"> </span><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank" style="color: blue; text-decoration: underline; ">https://puck.nether.net/mailman/listinfo/cisco-voip</a><span class="Apple-converted-space"> </span><br><br><br><span style="color: white; ">itevomcid</span><o:p></o:p></div></div></div></span></blockquote></div><br></div></body></html>