similar to: call center running Asterisk - sound quality-critical!

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

Except that mixmonitor still has a bug in it. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth

Yes. That's is the one. It is resolved now. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk-sound quality-critical! Wai Wu wrote: >

I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files "01/bemvindo" and "01/menu_top" are good.

Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on

Hi, I know this has already been discussed here, but I still have the problem even with 1.2.6: When I call a mailbox in a context "company" is doesn't play my busy message... It goes directly to the temp message... Am I doing something wrong? == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp("SIP/200.234.208.250-0840f548", "Voicemail de

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi, I am doing some tests with Asterisk's ACD capability, and as far as I could go I have realized that each agent defined in agents.conf must keep a session (call) open with Asterisk in order to be considered online. When a user calls, the agent receives a beep notification in his softphone and he answers to the pending call in the open channel and after the call ends he remains on the open

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["

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

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

Hi All, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can

A week or so ago, I explained that we need to "blank" our call recording when some sensitive information like credit cards where being discussed. With the lists help, I managed to find the pause/ unpause monitor commands. That works great. However (there is always a however), what that now means is that the length of the call does not match the length of the call recording, so adding

Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx

Open-Xchange Security Advisory 2019-12-13 ? Product: Dovecot IMAP/POP3 Server Vendor: OX Software GmbH ? Internal reference: DOV-3719 Vulnerability type: NULL Pointer Dereference (CWE-476) Vulnerable version: 2.3.9 Vulnerable component: push notification driver Report confidence: Confirmed Solution status: Fixed by Vendor Fixed version: 2.3.9.1 Researcher credits: Frederik Schwan, Michael

Open-Xchange Security Advisory 2019-12-13 ? Product: Dovecot IMAP/POP3 Server Vendor: OX Software GmbH ? Internal reference: DOV-3719 Vulnerability type: NULL Pointer Dereference (CWE-476) Vulnerable version: 2.3.9 Vulnerable component: push notification driver Report confidence: Confirmed Solution status: Fixed by Vendor Fixed version: 2.3.9.1 Researcher credits: Frederik Schwan, Michael