similar to: bug in 1.2.24

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"

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! 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

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.

Hi, sorry to bump this one but I still have this problem. The file is always created but is always zero size. This is the dial plan that records the call: exten = _0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID}) exten = _0[1-8]X.,2,MixMonitor(${CALLFILENAME}.gsm,b) The dial plan then calls a macro that makes the call. I?ve

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 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,

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, I want to start recording the caller automatically when the receptionist transfers a new sales lead to 567. I don't want the receptionist to have to press *1 manually for automon. Can someone recommend how best to accomplish this? exten => 567,1,Set(CALLERID(name)=SALES CALL) exten => 567,n,Playback(recorded-for-training) exten =>

I have an application where we receive calls on an inbound PRI. After hours, our Asterisk box dials our answering service on an outbound PRI and then bridges the caller to the answering service. The flow looks like this: (CALLER)INBOUND_PRI --> CONTEXT --> GOSUB(Incoming) --> GOSUB(bridge-to-anssrv) --> DIAL(answering_service) --> OUTBOUND_PRI(service) This has been working

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

I'm trying to set up basic monitoring for a specific extension (5004) to record all outgoing and incoming calls and save them as WAV files. I've set this in the extensions.conf file: exten => 5004,1,Answer exten => 5004,2,Wait,1 exten => 5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA LLERIDNUM}) exten => 5004,4,Monitor,wav|${CALLFILENAME} But it

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

Hi all, I'm trying to make a context that will monitor a call and when it's completed it would e-mail the wav to a specified mail adres. So I made a standard context that records a call, like this: exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ {TIMESTAMP}) exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m}) exten =>

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now