<div>1) Check directmedia option in sip. If enabled set it to no</div><div>2) Check NAT option and RTP debug in live scenario for any particular agent</div><div>3) if not solved yet, Where are your storing your mixmonitor recording? On any storage ? If yes, try to record on local harddisk.</div>
<div>4) Remove mixmonitor and test again</div><div> </div><div>Hope you find can find problem 99% in above scenario.</div><div> </div><div>Regards,</div><div> </div><div>Bharat Lalcheta<br></div><div class="gmail_quote">On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="ltr"><br><div class="gmail_extra"><div class="gmail_quote">
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn <span dir="ltr"><<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">Asterisk 11.1.0<br>
Various soft-phone SIP clients<br>
call center with 10-12 agents online at once using asterisk queue<br>
<br>
Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.<br>
<br>
Looking for ideas on how to begin to diagnose this or clues about what might be wrong.<br>
Is there a console command that will show details of a specific call in progress that might have some clues?<br>
<br>
-- <br>
<br>
Mitch<br>
<br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<u></u>mailman/listinfo/asterisk-<u></u>users</a><br>
</blockquote></div><br></div><div class="gmail_extra">Silly guess, If there is no then NAT did you check that your headphones work properly every time you start the softphone? This has happened to me in past.</div><span class="HOEnZb"><font color="#888888">
<div class="gmail_extra"><br></div><div class="gmail_extra">--Satish Barot</div><div class="gmail_extra">Ahmedabad, India.</div></font></span></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Bharat Lalcheta