search for: audiohook_inherit

I have a customer that needs to record all calls coming in and out. The problem I am having is when a call comes in to the operator and it is transferred to another extension. The first mixmonitor begins recording when the operator picks up but the recording stops when the call is transferred. I need to have a single recording for the complete call no matter how many times it is transferred.

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

...cro-apprecord] exten => s,1,Playback(beep) exten => s,n,Set(RECORDFILE=/var/spool/asterisk/autorecord/${STRFTIME(${EPOCH},,%Y/%m/%d/%H%M%S)}-${UNIQUEID}-^-${CALLERID(num)}) exten => s,n,Set(CDR(userfield)=${RECORDFILE}) exten => s,n,MixMonitor(${RECORDFILE}.wav) exten => s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => s,n,NoOp(CDR(userfield) = ${CDR(userfield)}) exten => s,n,MacroExit The NoOp shows the userfield is set correctly but the userfield is blank in my MySQL cdr database. I set CDR(userfield) elsewhere in the dialplan and this works so it seems to be related to being...

...e found out, but we want the second part of the conversation to be recorded in the queue_log as well, for stats reporting for reviews of employee performance. Is anyone aware of a relatively easy way of implementing this? Whether it's by a patch or something else? Basically something similar to audiohook_inherit, which we use to allow mixmonitor to continue recording the call after it's been transferred. I've looked around, but haven't found anything. Thanks. Cheers, Darrin Henshaw

...uot;MASTER_CHANNEL(ONETOUCH_REC)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:8] Set("SIP/1010-00000162", "MASTER_CHANNEL(REC_STATUS)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:9] Set("SIP/1010-00000162", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [s at macro-one-touch-record:10] MixMonitor("SIP/1010-00000162", "2012/08/21/out-7190000000-1010-20120821-183119-1345595479.530.wav,a,") in new stack == Begin MixMonitor Recording SIP/1010-00000162 -- Executing [s at mac...

Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the filename. However, what I am a little unsure about is the incoming number/called number/extension - passing that information on to part

...File=${recDir}/${STRFTIME(${EPOCH},,%y%m%d%H%M%S)}D${ARG1} N${ARG2}ID${UNIQUEID}.gsm) exten=>s,n(setacc),set(recFile=${recDir}/${CDR(linkedid)}.gsm) exten=>s,n(makedir),System(/bin/mkdir -p /var/spool/asterisk/monitor/${recDir}) exten=>s,n,MixMonitor(${recFile},a) exten=>s,n(finrec),Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME}) exten=>s,n,Set(__chanrecording=/var/spool/asterisk/monitor/${recFile}) exten=>s,n,NoOp(Recording to ${MIXMONITOR_FILENAME}) exten=>s,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel: ${CHANNEL},FileName: ${M...

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

...0103", "1?Set(__REC_POLICY_MODE=always)") in new stack -- Executing [out at sub-record-check:2] GosubIf("SIP/101-00000103", "1?record,1(exten,0149xxxxxx,101)") in new stack -- Executing [record at sub-record-check:1] Set("SIP/101-00000103", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [record at sub-record-check:2] MixMonitor("SIP/101-00000103", "2015/03/20/out-0149xxxxxx-101-20150320-164340-1426869820.301.wav,,") in new stack -- Executing [record at sub-record-check:3] Set("SIP/101-00000103",...

...0192", "1?Set(__REC_POLICY_MODE=always)") in new stack -- Executing [out at sub-record-check:2] GosubIf("SIP/300-00000192", "1?record,1(exten,0176XXXXXX,300)") in new stack -- Executing [record at sub-record-check:1] Set("SIP/300-00000192", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [record at sub-record-check:2] MixMonitor("SIP/300-00000192", "2015/03/27/out-0176XXXXXX-300-20150327-183519-1427481319.470.wav,,") in new stack -- Executing [record at sub-record-check:3] Set("SIP/300-00000192",...

...Set(__REC_POLICY_MODE=always)") in new stack > -- Executing [out at sub-record-check:2] GosubIf("SIP/101-00000103", > "1?record,1(exten,0149xxxxxx,101)") in new stack > -- Executing [record at sub-record-check:1] Set("SIP/101-00000103", > "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack > -- Executing [record at sub-record-check:2] > MixMonitor("SIP/101-00000103", > "2015/03/20/out-0149xxxxxx-101-20150320-164340-1426869820.301.wav,,") in > new stack > -- Executing [record at sub-record-check:3] Set(&quo...

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

...ot;) in new stack >>> -- Executing [out at sub-record-check:2] GosubIf("SIP/101-00000103", >>> "1?record,1(exten,0149xxxxxx,101)") in new stack >>> -- Executing [record at sub-record-check:1] Set("SIP/101-00000103", >>> "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack >>> -- Executing [record at sub-record-check:2] >>> MixMonitor("SIP/101-00000103", >>> "2015/03/20/out-0149xxxxxx-101-20150320-164340-1426869820.301.wav,,") in >>> new stack >>> -- Executing...

...t;) in new stack ... -- Executing [q at sub-record-check:1] GosubIf("Local/9991416445 at from-internal-0000017b;1", "1?recq,1(q,6001,9991416445)") in new stack -- Executing [recq at sub-record-check:1] Set("Local/9991416445 at from-internal-0000017b;1", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [recq at sub-record-check:2] Set("Local/9991416445 at from-internal-0000017b;1", "MONITOR_FILENAME=2015/04/16/q-6001-9991416445-20150416-175556-1429224932.21696") in new stack -- Executing [recq at sub-record-check:3] Mi...

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and write factory 0xae17e18 both fail to provide 160 samples [May 3

...] VERBOSE[9745] pbx.c: -- Executing [out at sub-record-check:2] GosubIf("SIP/1002-00000292", "1?record,1(exten,09999999999,1002)") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [record at sub-record-check:1] Set("SIP/1002-00000292", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [record at sub-record-check:2] MixMonitor("SIP/1002-00000292", "2013/09/25/out-09999999999-1002-20130925-135733-1380097653.1271.gsm,,") in new stack [2013-09-25 13:57:33] VERBOSE[9745...