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...@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...@claborn.net >> <mailto:mitch...@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 >> <satish4aster...@gmail.com >> <mailto:satish4asterisk@gmail.**com<satish4aster...@gmail.com> >> > >> <mailto:satish4asterisk@gmail.**__com >> <mailto:satish4asterisk@gmail.**com <satish4aster...@gmail.com>>>> >> wrote: >> >> >> On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn >> <mitch...@claborn.net <mailto:mitch...@claborn.net> >> <mailto:mitch...@claborn.net <mailto:mitch...@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
