[asterisk-users] Diagnosing call problem
Mitch Claborn
mitch_ml at claborn.net
Tue Mar 19 17:59:03 CDT 2013
The network is all on a single LAN segment - there is no NAT involved
anywhere. Agents do not have firewall or active anti-virus. See other
posts for a SIP trace.
Mitch
On 03/19/2013 12:45 PM, Bharat Lalcheta wrote:
> Firewall can cause problem on client side. Check antivirus or firewall
> on agent pc
> Please provide your network setup for getting better idea of problem
>
> On Mar 19, 2013 10:10 PM, "Mitch Claborn" <mitch_ml at claborn.net
> <mailto: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>
> <mailto: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>>
> <mailto:satish4asterisk at gmail.
> <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>>
> <mailto: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
>
>
> --
>
>
> _________________________________________________________________________________
> -- 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>
>
> <http://lists.digium.com/____mailman/listinfo/asterisk-____users
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users>>
>
>
> <http://lists.digium.com/____mailman/listinfo/asterisk-____users
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users>
>
> <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
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users>
>
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users
> <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
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users>
>
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users
> <http://lists.digium.com/mailman/listinfo/asterisk-users>>
>
>
> --
>
> _____________________________________________________________________________
> -- 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>
>
> <http://lists.digium.com/__mailman/listinfo/asterisk-__users
> <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
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
> --
> _________________________________________________________________________
> -- 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>
>
>
>
> --
> _____________________________________________________________________
> -- 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