hi satish, try to debug rtp on that ip and look rtp flow you can also test directmedia=no i encounter this as well i server is on public ip and clients connect via vpn , vpn server is also same asterisk server calls come in via public ip and go to call center via vpn i solved this by directmedia=no canreinvite=no
On Tue, Mar 19, 2013 at 5:51 AM, Satish Barot <[email protected]>wrote: > > On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn <[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> >> > > 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 >
-- _____________________________________________________________________ -- 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
