similar to: Recordings in the bank.

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

Greetings. Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work ok. Except for one thing. I have a call to MixMonitor. This is implementing a dictaphone kind of app. With forwarding recordings to email and storing them on the server. The process works so that we dial into Asterisk and answer the phone, initiate MixMontior and WaitExten until recording finishes. Problem is

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

I have a project here that involves streaming conversations out to an icecast server, and it would be great if asterisk were able to do this nicely. So far, I've got it working by using a simple dialplan like this: exten => 22,1,MixMonitor(test.wav) exten => 22,2,Dial(SIP/blabla@blabla.com) No problems at all if I record to a file, but then I made test.wav a fifo, and had oggenc read

Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the

Hi List, just to know if the issue is only a problem of mine or if it is a general issue due to the new MS Office pack. I'm using R 2.11.1 32 bits in a Windows 7 x64 with the MS office 2010 x64 installed. I can import .xls files normally (the same way I did with my Excel 2007 32 bits). But the function odbcConnectExcel2007 isn't able to import .xlsx files now that I have the new version of

i have installed the asterisk 1.6 before that installed the necessary packages in Debian, * i followed the steps as follows, root at astserver: ~# apt-get install unixodbc unixodbc-dev odbc-postgresql postgresql-8.1 postgresql-contrib postgresql-dev * then i installed the asterisk 1.6 version with the odbc modules as in the selected list. * then i created the database as asterisk and the user

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to wav >>>> (centos6) >>>> when the system is

Can you explain? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Tiago Geada <tiago.geada at gmail.com> </div><div>Date:03/07/2014 9:04 PM (GMT+02:00) </div><div>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re:

Hi Guys, Merry Christmas and Happy new Year. I am looking for some assistance from the group as i think this might already have been tried before. i have an asterisk server with a external USB Harddisk Drive, just to store recordings. I am using the mixmonitor application for doing the recordings. When i have active calls that are being recorded to the USB Drive, and if my USB disk fails for

Dne 28.1.2016 v 13:37 Brian :: napsal(a): > when you say load - how many concurrent calls? Is there transcoding > happening? sip / PRIs ? what load? > 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) > On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > >

I got this notion monitor-format = wav49 wav49 presents much louder than regular wav and gsm in my experience -- _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 22, 2010 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Hello, I am wondering has anyone used Live Recording (monitor or mixmonitor) on to Storage Server via network 1 Gigabit connection? Does it perform well, let say about 50 live recordings at the same time. I am planning to make some system changes at work. I would like to put Asterisk VM on a ESXi host and the datastore will be hosted on Storage Server. On a ESXi host, there will be a few

Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such

Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordings for no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2

> > Sorry, of course cdr.conf not queues.conf. marcus > > Am 01.12.2010 19:16 schrieb "marcus rothe" <synco16 at googlemail.com>: > > > Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? > Regards Marcus > > Thanks very much, I include the line "unansweredy=yes" in the cdr.conf and solve the problem. Thanks again! --