[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 08:09:23 CDT 2013
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>> 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
>
>
> --
> _________________________________________________________________________
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every
> Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/__mailman/listinfo/asterisk-__users
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
> 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:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Bharat Lalcheta
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list