similar to: Call hung up?

I had lines 3 and 4 and added line 1 and 2 to extensions.conf exten => 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,2,Monitor(wav,${CALLFILENAME},m) exten => 106,3,hint,SIP/106 exten => 106,4,Macro(stdexten,106,${HINT}) I received this warning: WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

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]

Hi, I wonder if anyone has any suggestions on how to setup a network to run VPN over wireless. I currently have: Wireless Laptop ----> Router with VPN pass through ---> DSL modem. There is also a wired desktop running Win '98se connected to the router. The Wireless Laptop is running XP Pro sp1. I am open for suggestions on how to run VPN over the wireless. I just want to protect

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 =>

I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 10000 and forwarded 10000-10400. Is there a possibility Express Talk won't work in the 10000 range? Is it possible to limit Asterisk to 8000-8020? Thank you, Gary

I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary

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 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 =>

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,

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

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 =>

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 =>

OK. I now have call recording working for both incoming and outgoing calls. Now I want to make those wavs into mp3. I could launch a script from cron that checks for new wavs and converts them. But that wouldn't be so elegant. Launching it from * on hangup would be nicer. How is it done? [outgoing] exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten =>

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

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

Does soxmix works with asterisk ver. 0.9? I have ver. sox-12.17.5 on Gentoo but the option "m" does not combine two WAV files (In and Out) into one file. I have two separate files in /monitor folder. exten => 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 711,2,Monitor(wav,${CALLFILENAME},m) exten => 711,3,Dial(${sales_support},20,r) exten =>

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