similar to: Voicemail volume adjustment

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port

Is there a way to edit the options available in the voicemail menu trees? My users are complaining that it's too complicated (I know, it's not really complicated), and I wanted to remove some of the options if this is possible. So far I havent' found any info on the wiki or searches, not that it isn't out there.. I just cant' seem to find it.. Any pointers? Thanks Dan

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

On Nov 05 11:32:49, hans at stare.cz wrote: > On Nov 05 11:05:34, hans at stare.cz wrote: > > > Did you also try to listen at the beginning, shortly before the real tone appears in the audible spectrum? While significantly larger, Opus had produced significant ghost noise (much less than Vorbis did)... I experience the "same" low level noise even in a wav file, even on

I have a ring group set up with 3 extensions. we'll use 14, 15 and 16. When a call comes in, it rings all three extensions. If one particular extension already is on the phone, it completely skips that phone and only rings the other 2. Example to explanation sake is: Call comes in, ext. 14 is already in the middle of a call, 15 and 16 will ring normally, but 14 does not have any

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

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

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

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

Hello. Me again. Have you tried to encode piano solo? Noticed high bitrate Opus gave? And there's also artefact at 15kHz which wasn't in the original audio. Visible with Spek program. Download FLAC and Opus both files, new link: http://www.filedropper.com/example_3 FLAC full: 1084 kbps; FLAC solo: 465 kbps. with --bitrate 160: Opus full: 158 kbps; Opus solo: 190 kbps. Included also Spek

We are switching from Windows Media to Icecast. I want to get as close to open source as possible. That means that I need the following open-source items: Player for Windows (not Winamp if possible - it's now owned by AOL.) Player for Mac Player for Linux We will also be podcasting, so that means our encoding needs to be compatible with the Apple iPod. Here's what it supports: MP3

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

How do I combined these in and out wav files on the fly through asterisk to where I hear the whole conversation and only have one wav-file (i.e. : agent-1001-asterisk-478-1127389080-17-in_out.wav) agent-1001-asterisk-478-1127389080-17-in.wav agent-1001-asterisk-478-1127389080-17-out.wav __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

Hi, It is important for some applications (voicemail and the like) to control play volume and speed if instructed by the caller. The question is: how to do it the right way with asterisk. As far as volume - adding txgain and rxgain to the call structure[s] and setting gains from there (instead of taking global values from config files) looks like an OK solution. As far as speed - the easy way