hi satish,<div>try to debug rtp on that ip and look rtp flow you can also test directmedia=no i encounter this as well i server is on public ip and clients connect via vpn , vpn server is also same asterisk server calls come in via public ip and go to call center via vpn i solved this by directmedia=no canreinvite=no <br>
<br><div class="gmail_quote">On Tue, Mar 19, 2013 at 5:51 AM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</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"><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 class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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>
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</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>
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