similar to: Queue Monitoring

Hi guys, I'm running Asterisk 1.2.13 on a Debian Linux system (that was just the version that was packaged for it). I've been using monitor() to record calls, with fairly satisfactory results - at least until the last few months. I've been recording VoIP calls, and using monitor() with no arguments, so I'm getting separate wav files for each leg (both use ALAW, BTW), and

[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

Hi, I have started using the call recording facilities in Asterisk 1.2 recently, and having worked out some of the foibles regarding call forwarding etc etc, I think I have a mostly working system. I do still seem to have a problem with recording volume though. It seems that all SIP call legs are recorded at "normal" volume, but all my Zap (ISDN) and IAX (via Provider -> ISDN) calls

Hi, I was just wondering if it is possible to cause a script to run (configured in the domu.cfg file) each time "xm create domu.cfg" is run, but before the machine is actually started? ie, I''d like to "setup" the disks for the VM before xen/qemu tries to use the devices and allocate to the domu. I''m using xen 4.1.3 on Debian testing, and using the xm

I''ve been trying to migrate a win nt 4 machine to a xen domu for the past few months with no success. However, on my current attempt, the original hardware no longer boots, so I''m trying to resolve the issues with xen properly, or else take a long holiday... Anyway, the physical machine had a 9G drive (OS drive), a 147 G drive (not in use) and a 300G drive (all SCSI Ultra320 on

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

I have been happily using NUT to monitor a couple of Eaton UPS. These are independent UPSs, each driving one "leg" of the power in our data center. In this scenario, I've told my clients they need to have one UPS working and to shutdown when the last UPS reaches critical battery. I have recently learned that our power arrangement will be changing to a parallel-fed/redundant

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension when it's finished. The second leg is being initiated by a .call file like:

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Hello, If I connect an Asterisk 1.6 to a PBX via Q.SIG and A (on the PBX) calls B (a SIP phone on Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ? The Q.SIG "Path Replacement Feature" requires the following: After both legs of the

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

Hi All, In previous mail lists, people talked about a solution to record large amount of simultaneous calls. And then it seems that RAM disk solution was the best choice due to the I/O bottleneck of Hard disk (System). Please find the previous discussion as follows: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.htm l

> -----Original Message----- > From: Adam Goryachev [mailto:mailinglists@websitemanagers.com.au] > Sent: September 7, 2004 8:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap > interface? {clip} > Have you considered adding the r option to the Dial command, so they > might hear ringing

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel

I found this property in queue.conf ; Calls may be recorded using Asterisk's monitor resource ; This can be enabled from within the Queue application, starting recording ; when the call is actually picked up; thus, only successful calls are ; recorded, and you are not recording while people are listening to MOH. ; To enable monitoring, simply specify "monitor-format"; it will be

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > >

I gave a queue setup like this, but I also have it setup so that if no agents are online, the caller cannot get in but I discovered that if that's the case, the call hangsup on the caller: [soportetecnico] ;Soporte Tecnico exten => 2,1,Playback(${SONIDOS}/transferringcall) exten => 2,2,Queue(Soporte-Tecnico) exten => 2-.,1,Playback(noagents) I want to play a message tothecaller