<p>Firewall can cause problem on client side. Check antivirus or firewall on agent pc <br>
Please provide your network setup for getting better idea of problem</p>
<div class="gmail_quote">On Mar 19, 2013 10:10 PM, "Mitch Claborn" <<a href="mailto:mitch_ml@claborn.net">mitch_ml@claborn.net</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
rtp debug on the calls that do not work correctly shows packets from server to client only, none from client to server.<br>
<br>
I do have<br>
<br>
nat=no<br>
directmedia=no<br>
<br>
in sip.conf. Are there other settings that might apply?<br>
<br>
This last instance that I looked at, the problem persisted even after restarting the client softphone program. It was fixed after rebooting the client computer.<br>
<br>
Any ideas on a next step for debugging? I was thinking I would start a wireshark trace to see if the rtp packets are actually leaving the client computer.<br>
<br>
<br>
<br>
Mitch<br>
<br>
On 03/19/2013 08:28 AM, Bharat Lalcheta wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
rtp set debug ip 1.2.3.4<br>
where 1.2.3.4 is ip of your particular agent.<br>
Say your x agent is not getting voice, rtp debu his ip.<br>
You got rtp packet from and to for that ip. If you find rtp packet from<br>
your agent to your server ip and rtp packet from your server to agent<br>
ip, then no need to check anything in asterisk. Its related to your<br>
agent pc problem<br>
If you find any single side rtp, then its problem related to nat or<br>
direct media etc.<br>
if mix monitor is on storage than only you can face problem and thats<br>
also very rare. In that case you get voice in break, but it will be from<br>
both side not in single side. So, this is not your problem at all.<br>
Hope you will get something in rtp debug.<br>
R u using any trunk then also check rtp debug between your server and trunk<br>
regards,<br>
<br>
Bharat Lalcheta<br>
<br>
<br>
On Tue, Mar 19, 2013 at 6:39 PM, Mitch Claborn <<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a><br>
<mailto:<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a>>> wrote:<br>
<br>
Thanks for the suggestions.<br>
<br>
1) directmedia was taking the default of "yes". I set to "no".<br>
Will watch and see.<br>
<br>
2) NAT is turned off (nat=no). I've never done any RTP debugging.<br>
Is that "rtp set debug on ip 1.2.3.4"? How would I interpret the<br>
output?<br>
<br>
3) mixmonitor recordings are stored on a local disk (RAID array,<br>
very fast)<br>
<br>
4) This would have to be a last resort option, as there is a<br>
business requirement to record the agent calls<br>
<br>
<br>
Mitch<br>
<br>
On 03/19/2013 12:01 AM, Bharat Lalcheta wrote:<br>
<br>
1) Check directmedia option in sip. If enabled set it to no<br>
2) Check NAT option and RTP debug in live scenario for any<br>
particular agent<br>
3) if not solved yet, Where are your storing your mixmonitor<br>
recording?<br>
On any storage ? If yes, try to record on local harddisk.<br>
4) Remove mixmonitor and test again<br>
Hope you find can find problem 99% in above scenario.<br>
Regards,<br>
Bharat Lalcheta<br>
<br>
On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot<br>
<<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a> <mailto:<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.<u></u>com</a>><br>
<mailto:<a href="mailto:satish4asterisk@gmail." target="_blank">satish4asterisk@gmail.</a><u></u>__com<br>
<mailto:<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.<u></u>com</a>>>> wrote:<br>
<br>
<br>
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn<br>
<<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a> <mailto:<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a>><br>
<mailto:<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a> <mailto:<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a>>><u></u>> wrote:<br>
<br>
Asterisk 11.1.0<br>
Various soft-phone SIP clients<br>
call center with 10-12 agents online at once using<br>
asterisk queue<br>
<br>
Occasionally an agent will get a call (or more often a<br>
series of<br>
calls in a row) where neither party can hear the other,<br>
or can<br>
only hear each other sporadically. A MixMonitor<br>
recording of<br>
the call plays only the caller - none of the agent's<br>
audio is<br>
heard in the recording.<br>
<br>
Looking for ideas on how to begin to diagnose this or clues<br>
about what might be wrong.<br>
Is there a console command that will show details of a<br>
specific<br>
call in progress that might have some clues?<br>
<br>
--<br>
<br>
Mitch<br>
<br>
<br>
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<br>
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<br>
<br>
Silly guess, If there is no then NAT did you check that your<br>
headphones work properly every time you start the<br>
softphone? This<br>
has happened to me in past.<br>
<br>
--Satish Barot<br>
Ahmedabad, India.<br>
<br>
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