[asterisk-users] Diagnosing call problem

[Asghar Mohammad]

hi satish,
try to debug rtp on that ip and look rtp flow you can also test
directmedia=no i encounter this as well i server is on public ip and
clients connect via vpn , vpn server is also same asterisk server calls
come in via public ip and go to call center via vpn i solved this by
directmedia=no canreinvite=no

On Tue, Mar 19, 2013 at 5:51 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.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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