[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 11:39:59 CDT 2013
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 <mitch_ml at claborn.net
> <mailto: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>
> <mailto:satish4asterisk at gmail.__com
> <mailto: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>
> <mailto: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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