similar to: Auto recording calls?

I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

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,

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 =>

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

How do I include a file (not a context) in AEL? #include "filename" returns an error. Thanks, Jay

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]

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial

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 =>

Alex thank you for your response. In this case we are USING INBAND, though I have tried both. Nothing works. Yes ser is configured with mediaproxy. Thank you, -JK JK, In-band or RFC2833 DTMF signaling? Also, unless you have SER configured with a media proxy, the actual "call" is not running through SER. It's a signaling proxy only. -- Alex Balashov Evariste Systems Web :

Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were "left over" from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash

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

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",

[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

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 =>

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 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 =>