Thanks for the input. I think that works as my other recordings work. I will test that again regardless.
Is there no real other way to know why MixMonitor fails or look more into it? Regards, Bruce On Wed, May 4, 2011 at 5:03 AM, salaheddine elharit < [email protected]> wrote: > hi > > you can add this in extenssion.conf > > > exten => 223,1,Answer() > > exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0)) > > exten => 223,3,Dial(SIP/223) > > exten => 223,4,Hangup() > > i can record without any issue in /var/spool/asterisk/monitor > > > 2011/5/4 Bruce B <[email protected]> > >> Thanks for the input. >> >> Yes, I did call out many times, but the recording doesn't happen even >> after the call is bridged and there is two way audio. I also took out the >> "b" option and so it should recording the ringing right (even before call is >> bridged) but it doesn't do that or any recording at all. >> >> Any other suggestions as to what I can do to see why this is not >> recording? >> >> Regards, >> >> >> On Tue, May 3, 2011 at 2:13 AM, virendra bhati <[email protected]>wrote: >> >>> Hi, >>> >>> As per your Dialplan MixMonitor will work after call bridge, In you case >>> still call is not bridge. That's why MixMonitor is waiting of call bridge... >>> >>> * >>> MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,) >>> option b=>** A bridge flag allows recording to only take place when the >>> channel is bridged.* >>> >>> So just for test make sip call and start mixmonitor to test the recorded >>> file. >>> default path od recording id >>> * >>> /var/spool/asterisk/monitor/ >>> >>> * >>> On Tue, May 3, 2011 at 10:40 AM, Bruce B <[email protected]> wrote: >>> >>>> Hi everyone, >>>> >>>> For some reason MixMonitor doesn't record when it should; It actually >>>> shows the MixMonitor line just fine on the CLI. How can MixMonitor be >>>> debugged for things like privilege issues or filename issues? >>>> >>>> **I had this working at one point and then stopped working. Not sure >>>> what I changed. >>>> >>>> System Info: >>>> Asterisk 1.4.21.2 >>>> Queuemetrics 1.6.3.0 >>>> >>>> >>>> [queuedial] >>>> ; this piece of dialplan is just a calling hook into the >>>> [qm-queuedial] context that actually does the >>>> ; outbound dialing - replace as needed - just fill in the same >>>> variables. >>>> exten => _XXX.,1,Set(QDIALER_QUEUE=q-${EXTEN:0:3}) >>>> exten => _XXX.,n,Set(QDIALER_NUMBER=${EXTEN:3}) >>>> exten => _XXX.,n,Set(QDIALER_AGENT=Agent/${CALLERID(num)}) >>>> exten => _XXX.,n,Set(QDIALER_CHANNEL=ZAP/g0/${QDIALER_NUMBER}) >>>> exten => _XXX.,n,Set(QueueName=${QDIALER_QUEUE}) >>>> *exten => _XXX.,n,MixMonitor(Q-${QDIALER_QUEUE}-${UNIQUEID}.WAV,b,)* >>>> exten => _XXX.,n,Goto(qm-queuedial,s,1) >>>> >>>> CLI output: >>>> -- Called 4904166356574@queuedial/n >>>> -- Executing [4904166356574@queuedial:1] >>>> Set("Local/4904166356574@queuedial-d851,2", "QDIALER_QUEUE=q-490") in >>>> new stack >>>> -- Executing [4904166356574@queuedial:2] >>>> Set("Local/4904166356574@queuedial-d851,2", >>>> "QDIALER_NUMBER=4166356574") in new stack >>>> -- Executing [4904166356574@queuedial:3] >>>> Set("Local/4904166356574@queuedial-d851,2", >>>> "QDIALER_AGENT=Agent/19053640558") in new stack >>>> -- Executing [4904166356574@queuedial:4] >>>> Set("Local/4904166356574@queuedial-d851,2", >>>> "QDIALER_CHANNEL=ZAP/g0/4166356574") in new stack >>>> -- Executing [4904166356574@queuedial:5] >>>> Set("Local/4904166356574@queuedial-d851,2", "QueueName=q-490") in new >>>> stack >>>> * -- Executing [4904166356574@queuedial:6] >>>> MixMonitor("Local/4904166356574@queuedial-d851,2", >>>> "Q-q-490-1304399098.18.WAV|b|") in new stack* >>>> -- Executing [4904166356574@queuedial:7] >>>> Goto("Local/4904166356574@queuedial-d851,2", "qm-queuedial|s|1") in new >>>> stack >>>> -- Goto (qm-queuedial,s,1) >>>> >>>> Trying to locate file: >>>> root@pbx:~ $ updatedb >>>> root@pbx:~ $ locate Q-q-490-1304399098.18.WAV >>>> root@pbx:~ $ ls /var/spool/asterisk/monitor/Q-q* >>>> ls: /var/spool/asterisk/monitor/Q-q*: No such file or directory >>>> >>>> I also turned on the Debug but I couldn't see anything out of the norm. >>>> As you can see above the CLI output is just fine. >>>> >>>> Thanks, >>>> Bruce >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> >>> -- >>> >>> >>> >>> ----- >>> Thanks and regards >>> >>> Virendra Bhati >>> +91-9172341457 >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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
