[asterisk-users] Diagnosing call problem

[Bharat Lalcheta]

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 <satish4asterisk at gmail.com>wrote:

>
> On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn <mitch_ml at claborn.net>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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-- 
Bharat Lalcheta
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