similar to: Call recording format

For some reason we are seeing "Avoiding deadlock for channel" in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are "Avoiding deadlock". What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's.

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

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

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

This is probably the most eternal question concerning rsync, but I'm really in doubt since I can't find the appropriate answer or solution anywhere for many months. I'm using rsync for simple daily back up data from one HD to another. It takes about 10 minutes daily under "nice -n 19". The problem is well-known as I see - several other services such as

> Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings If you're up to writing a bit of shell script, and are running on Linux, you could automate the conversion process so that it happens as soon as the recording is completed. Look at the

I am, as this question will make very obvious, not exactly a guru when it comes to audio under Linux. I'd like to record some prompts using a nice microphone and my Ensoniq sound card, and then convert them to gsm/wav for use with asterisk. What Linux command would I use to a) do the recordings, and then b) convert them to the appropriate formats? I know there are

Does anybody have useful tips on creating good quality recordings for use with prompts in asterisk? I'm interested in hearing input on hardware (mics, dats, sound cards, etc) and software (recording software, dsp) as well as recording techniques. Jerimiah Tularosa Communications

List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

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]

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google (

Hi, I''m pretty new to Rspec, so hope someone can help me out with this. I''m leveraging a couple pre-existing applications in my current project, both have fairly robust test suites in Rspec already. However, one is using Rspec''s builit in mocking, and the other is using mocha. If I set config.mock_with mocha in spec_helper.rb, then all of the rspec mocks fail. If I

Modified httf.conf file and added : ------------------------------------------------------ Alias /recordings/ "/var/spool/asterisk/monitorDONE/" <Directory "/var/spool/asterisk/monitorDONE"> Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all </Directory> Created a folder under vicidial as recordings. FULL_RECORDING is also enabled.

https://bugzilla.samba.org/show_bug.cgi?id=3491 Summary: throttle disk IO during filelist/directory parsing Product: rsync Version: 2.6.4 Platform: All URL: http://vilius.multiply.com/video/item/10 OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

Hi, I tried to use Monitor(wav,filename) function in dialplan to record Meetme conference. When I monitored on IAX2 or SIP channels in that conference It recorded all audio (in and out) but when I monitored on ZAP channels I could hear only IN audio and piece of OUT audio (announcement get pin and than nothing). Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package) and

[Cross-posted to the asterisk forums, so if it seems familiar...] My company is looking to implement 100% recording of inbound and outbound calls. We're pondering using Asterisk as a trunk-side monitoring system sitting between our Avaya Definity switch and our mixed bag of 6 T1 and PRI circuits. I've read with interest Matt Roth's thread on the Asterisk mailing list, however, since

Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the "wav" format are about twice the level of those made using "WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed and about 10 times the size of the other formats. -brian