[asterisk-users] Diagnosing call problem

[Bharat Lalcheta]

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 <mitch_ml at claborn.net> 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
>> <satish4asterisk at gmail.com <mailto:satish4asterisk at gmail.**com<satish4asterisk at gmail.com>>>
>> wrote:
>>
>>
>>     On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
>>     <mitch_ml at claborn.net <mailto: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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-- 
Bharat Lalcheta
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