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]>> wrote:
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
<[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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