<div dir="ltr">Try setting the channel var record_waste_resources=true in the dp before calling VM or globally in vars.xml<div style>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....</div>
<div style> </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Apr 19, 2013 at 11:35 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div>This is interesting. I suspect she has given you only the facts as she sees them and not necessarily all of the facts.<br>
</div>I believe you'll need pcaps of the calls in question to know for sure.<br>
<br></div>As far as FreeSWITCH being able to tell the difference between a human and a voicemail/answering machine... not without a special module. :)<br><br></div>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.<br>
<br></div>-MC <br></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Fri, Apr 19, 2013 at 7:55 AM, Sean Devoy <span dir="ltr"><<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div bgcolor="white" background="cid:image001.gif@01CE3CEC.36DFFB40" link="#1F86FF" vlink="#005DC9" lang="EN-US">
<img src="cid:image001.gif@01CE3CEC.36DFFB40" style="width:0;min-height:0" height="0" width="0"><div>
<p class="MsoNormal"><span style="color:#1f497d">Hi All,<u></u><u></u></span></p><p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><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.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><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!<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><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.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><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.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:#1f497d">The log just shows the call being answered … then Normal Clearing.<u></u><u></u></span></p><p class="MsoNormal">
<span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><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.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:#1f497d">Thanks in advance.<span><font color="#888888"><u></u><u></u></font></span></span></p>
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