similar to: MixMonitor Files Always Empty

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

Is there an equivalent stopmonitor command if you are using MixMonitor ? StopMonitor does not seem to have an effect on MixMonitor Julian.

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Hi all Using asterisk 16.25 I was trying to stop Mixmonitor using features. The code is executed but I realized that I was executing StopMixmonitor from another channel so I opted to use AMI. When I call MixMonitor I store the channel name in a var and then I use StopMixmonitor from AMI sending the stored channel name as parameter. What I've seen is that the app returns failure and going

I need help need recording all call for my pbx but i am a novato in asterisk my confi for record is: exten=>_NXXXXXXXXX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) exten => _NXXXXXXXXX,n,MixMonitor(${CALLFILENAME}.gsm,m) exten => _NXXXXXXXXX,n,Dial(${TRUNK_CLIENTE}/${EXTEN}) -- Bayardo S?nchez Garc?a Web Developer - Internet

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

Hello list, how does StopMixMonitor know which 'monitoring channel' to stop when there are multiple conversations that are being monitored/recorded ?? I want to use StopMixMonitor in a macro, called from within applicationmap (features.conf). Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>

GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to somebody else outside the queue, of course... the bug comes in here.. Im using mixmonitor to record

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards!

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel