rtp debug on the calls that do not work correctly shows packets from
server to client only, none from client to server.
I do have
nat=no
directmedia=no
in sip.conf. Are there other settings that might apply?
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.
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.
Mitch
On 03/19/2013 08:28 AM, Bharat Lalcheta wrote:
rtp set debug ip 1.2.3.4
where 1.2.3.4 is ip of your particular agent.
Say your x agent is not getting voice, rtp debu his ip.
You got rtp packet from and to for that ip. If you find rtp packet from
your agent to your server ip and rtp packet from your server to agent
ip, then no need to check anything in asterisk. Its related to your
agent pc problem
If you find any single side rtp, then its problem related to nat or
direct media etc.
if mix monitor is on storage than only you can face problem and thats
also very rare. In that case you get voice in break, but it will be from
both side not in single side. So, this is not your problem at all.
Hope you will get something in rtp debug.
R u using any trunk then also check rtp debug between your server and trunk
regards,
Bharat Lalcheta
On Tue, Mar 19, 2013 at 6:39 PM, Mitch Claborn <[email protected]
<mailto:[email protected]>> wrote:
Thanks for the suggestions.
1) directmedia was taking the default of "yes". I set to "no".
Will watch and see.
2) NAT is turned off (nat=no). I've never done any RTP debugging.
Is that "rtp set debug on ip 1.2.3.4"? How would I interpret the
output?
3) mixmonitor recordings are stored on a local disk (RAID array,
very fast)
4) This would have to be a last resort option, as there is a
business requirement to record the agent calls
Mitch
On 03/19/2013 12:01 AM, Bharat Lalcheta wrote:
1) Check directmedia option in sip. If enabled set it to no
2) Check NAT option and RTP debug in live scenario for any
particular agent
3) if not solved yet, Where are your storing your mixmonitor
recording?
On any storage ? If yes, try to record on local harddisk.
4) Remove mixmonitor and test again
Hope you find can find problem 99% in above scenario.
Regards,
Bharat Lalcheta
On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot
<[email protected] <mailto:[email protected]>
<mailto:satish4asterisk@gmail.__com
<mailto:[email protected]>>> wrote:
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>> wrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using
asterisk queue
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.
Looking for ideas on how to begin to diagnose this or clues
about what might be wrong.
Is there a console command that will show details of a
specific
call in progress that might have some clues?
--
Mitch
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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.
--Satish Barot
Ahmedabad, India.
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