<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)"><!--[if !mso]><style>v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
</style><![endif]--><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Verdana;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.hoenzb
{mso-style-name:hoenzb;}
span.EmailStyle18
{mso-style-type:personal-reply;
font-family:"Verdana","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-family:"Calibri","sans-serif";}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'>Anthony – I think you got it backwards. This is a FS user calling out of FS and getting voicemail elsewhere, like Sprint. <o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'>In follow up she is “sure” the audio cut off in the middle of the callee’s outgoing message. Other users at that site with the same phones report the same problem,<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'>I have called the same number with a cisco phone from a different external site and do not have the problem. I have sent are a list of tests to see if we can reproduce the problem. Then I will pastebin mountains of data for you guys.<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'>My current hypothesis is some noise in their office environment caused the b-led audio on her phone to cut out in favor of a-led audio momentarily. Then when it returned to b-led audio, the remote end was already past the “beep” and recording. That could be why it is specific to voicemail calls. I know some speaker phones can do that, but could Polycom 335 handsets do it too?<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","sans-serif";color:#1F497D'>Thank you again for your time and efforts.<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Verdana","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"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Anthony Minessale<br><b>Sent:</b> Friday, April 19, 2013 5:53 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Audio Lost after 1 min<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Try setting the channel var record_waste_resources=true in the dp before calling VM or globally in vars.xml<o:p></o:p></p><div><p class=MsoNormal>This will cause useless silent RTP to be fired at the caller while they are recording. Without this it logically stops sending while its recording the message since you don't need to hear anything while you are recording your message. Some devices panic with no media and hangup thus this workaround....<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Fri, Apr 19, 2013 at 11:35 AM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<o:p></o:p></p><div><div><div><div><div><p class=MsoNormal>This is interesting. I suspect she has given you only the facts as she sees them and not necessarily all of the facts.<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'>I believe you'll need pcaps of the calls in question to know for sure.<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'>As far as FreeSWITCH being able to tell the difference between a human and a voicemail/answering machine... not without a special module. :)<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'>Get those pcaps and FS logs and see if there are any clues. If you can compare a working call against a non-working call then that might help you narrow down where to look.<o:p></o:p></p></div><p class=MsoNormal>-MC <o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><div><div><p class=MsoNormal>On Fri, Apr 19, 2013 at 7:55 AM, Sean Devoy <<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>> wrote:<o:p></o:p></p></div></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><div><p class=MsoNormal><img border=0 width=31 height=31 id="_x0000_i1025" src="cid:image001.gif@01CE3DBE.37718470"><o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>Hi All,</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>I have a report by a customer at a site that we had considerable trouble with NAT issues. I thought everything was now stable, but it sounds like it is not.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>She reports that she is only telling me now, but for some time when they call someone and get no answer and get directed to voicemail (the latest was a Sprint number) the audio from the remote answering machine cuts off before then end of the message. She did point out the call timer on her phone continued normally. She swears this does not happen when talking to a live person!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>First, I don’t think FS can even tell if the b-leg was answered by voice mail differently from a person. Please correct me if that is incorrect.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>So, I am proceeding on the assumption that calls that are answered take less than 60 seconds! 45 seconds of the reported call were waiting for Sprint to give up on the cell phone and route to voicemail.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>The log just shows the call being answered … then Normal Clearing.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>I tend to always suspect NAT or crappy Polycom 335s for this customer as that has been all my issues with them so far. Is there a NAT issues relating to max call length, etc? No other users (all cisco phones) on the same FS at other sites report this issue.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>Thanks in advance.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='color:#1F497D'>Sean</span><span style='color:#888888'><o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal style='margin-bottom:12.0pt'>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></blockquote></div><p class=MsoNormal style='margin-bottom:12.0pt'><span style='color:#888888'><br><br clear=all><br><span class=hoenzb>-- </span><br><span class=hoenzb>Michael S Collins</span><br><span class=hoenzb>Twitter: @mercutioviz</span><br><span class=hoenzb><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a></span><br><span class=hoenzb><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a></span><br><span class=hoenzb><a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a></span></span><o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></div><p class=MsoNormal><br><br clear=all><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>Twitter: <a href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:+19193869900 <o:p></o:p></p></div></div></body></html>