Grzegorz Pycia
2012-Oct-20 21:49 UTC
[asterisk-users] monitor application, file name change on attended transfer
Hi I have some problem with monitor application when call i transferred in attended mode and the transfer occurs before call is answered. Here is how it looks: A calls ----> B(let's assume ${UNIQUEUEID}=1) exten => _XXXX,1,NoOp seme => n,Set(MONITOR_FILENAME=call-${UNIQUEID}) same => n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm) When B answers the call, files call-1-in* and call1-out* are created. During The call, B tries to make attended transfer A is put on hold and B calls C using the same dialplan logic: B calls ----> C(let's assume ${UNIQUEUEID}=2) At the time off invoking monitor application none off the call-2 channels are monitored so the monitor application starts without errors, if B waits till C answers, everything is OK monitor starts recording and files call-2-in* and call-2-out* are created, When B transfers the call call-2 monitor is stopped. And call-2 files contain only the call between B and C. But there is problem when B does not wait until C answers the call, if transfer is done before C answers the call, the call-2* are not created and the call is still recorded to the call-1* files, but when the transferred call between A and C ends, the call-1* files get renamed to call-2* and the MONITOR_EXEC application is called with call-2* file names as parameters. This makes it impossible to locate the call record since the file names get changed, can someone tell if I should file a BUG report or is it intended to act like this? Regards
Binan AL Halabi
2012-Oct-22 01:14 UTC
[asterisk-users] monitor application, file name change on attended transfer
Hi, You are using b flag in monitor command. This means don't begin recording untill call is bridged. So what you get if you delete this flag ? // Binan ________________________________ Fr?n: Grzegorz Pycia <grzegorz.pycia at thulium.pl> Till: asterisk-users at lists.digium.com Skickat: l?rdag, 20 oktober 2012 23:49 ?mne: [asterisk-users] monitor application, file name change on attended transfer Hi I have some problem with monitor application when call i transferred in attended mode and the transfer occurs before call is answered. Here is how it looks: A calls ----> B(let's assume ${UNIQUEUEID}=1) exten => _XXXX,1,NoOp seme => n,Set(MONITOR_FILENAME=call-${UNIQUEID}) same => n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm) When B answers the call, files call-1-in* and call1-out* are created. During The call, B tries to make attended transfer A is put on hold and B calls C using the same dialplan logic: B calls ----> C(let's assume ${UNIQUEUEID}=2) At the time off invoking monitor application none off the call-2 channels are monitored so the monitor application starts without errors, if B waits till C answers, everything is OK monitor starts recording and files call-2-in* and call-2-out* are created, When B transfers the call call-2 monitor is stopped. And call-2 files contain only the call between B and C. But there is problem when B does not wait until C answers the call, if transfer is done before C answers the call, the call-2* are not created and the call is still recorded to the call-1* files, but when the transferred call between A and C ends, the call-1* files get renamed to call-2* and the MONITOR_EXEC application is called with call-2* file names as parameters. This makes it impossible to locate the call record since the file names get changed, can someone tell if I should file a BUG report or is it intended to act like this? Regards -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121021/ca865bdd/attachment.htm>
Jonathan Rose
2012-Oct-22 15:21 UTC
[asterisk-users] monitor application, file name change on attended transfer
Grzegorz Pycia wrote:> Hi > > I have some problem with monitor application when call i transferred > in > attended mode and the transfer occurs before call is answered. > > Here is how it looks: > > A calls ----> B(let's assume ${UNIQUEUEID}=1) > > exten => _XXXX,1,NoOp > seme => n,Set(MONITOR_FILENAME=call-${UNIQUEID}) > same => > n,monitor(alaw,/var/spool/asterisk/monitor/${MONITOR_FILENAME},bm) > > When B answers the call, files call-1-in* and call1-out* are created. > During The call, B tries to make attended transfer A is put on hold > and > B calls C using the same dialplan logic: > > B calls ----> C(let's assume ${UNIQUEUEID}=2) > > At the time off invoking monitor application none off the call-2 > channels are monitored so the monitor application starts without > errors, > if B waits till C answers, everything is OK monitor starts recording > and > files call-2-in* and call-2-out* are created, When B transfers the > call > call-2 monitor is stopped. And call-2 files contain only the call > between B and C. > > But there is problem when B does not wait until C answers the call, > if > transfer is done before C answers the call, the call-2* are not > created > and the call is still recorded to the call-1* files, but when the > transferred call between A and C ends, the call-1* files get renamed > to > call-2* and the MONITOR_EXEC application is called with call-2* file > names as parameters. > > This makes it impossible to locate the call record since the file > names > get changed, can someone tell if I should file a BUG report or is it > intended to act like this? > > RegardsAre you using Asterisk 1.8 or higher? A good way to mitigate this would be to use MixMonitor. It applies as an audiohook which should persist through transfers like the one you described, so you would just need to set AUDIOHOOK_INHERIT for MixMonitor in order to use it that way. One difference with this approach though would be that MixMonitor will automatically mix audio from both ends of the call into a single recording. That behavior can be worked around starting with Asterisk 10 by using the r and t options. I guess it's worth noting that if you aren't using 1.8 or higher there isn't really any point in filing a bug report since earlier versions aren't supported anymore. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com & http://asterisk.org