hi, rtp set debug ip 1.2.3.4 On Tue, Mar 19, 2013 at 2:09 PM, Mitch Claborn <[email protected]> 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 >> <[email protected] >> <mailto:satish4asterisk@gmail.**com<[email protected]>>> >> wrote: >> >> >> On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn >> <[email protected] <mailto:[email protected]>> 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> >> > >> >> >> 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> >> >> >> >> >> -- >> 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
