[asterisk-users] Diagnosing call problem

Tue Mar 19 12:02:49 CDT 2013

witch softphone you are using? on client pc installed some kind of
virtualpc like vmware or virtualbox? client pc have more then one network
interfaces?
you can capture sip invites from soft phone by enabling debug on client ip
sip set debug ip "ip of softphon" upload sip trace then somebody can halp
you, should provide more information's.

On Tue, Mar 19, 2013 at 5:39 PM, Mitch Claborn <mitch_ml at claborn.net> wrote:

> 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<satish4asterisk at gmail.com>
>> >
>>         <mailto: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>
>>         <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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>>
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>>
>>
>>
>>         --
>>         Bharat Lalcheta
>>
>>
>>
>>
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>>
>>
>>
>> --
>> Bharat Lalcheta
>>
>>
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