similar to: controlling recording

Hi All, I've been playing about with the RECORD FILE agi function and am finding two distinct problems with the resulting wav file when using a non zero sample offset. Specifically, I call the function with a zero offset and a given filename (the "original" recording), and then later call it with the same filename and a non-zero offset (the "overdub"). When I do this, I

Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav"

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

On 06/22/2015 07:20 PM, Frank Cox wrote: > The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. > > Now I want to record a "recitation", which consists of me playing my piano and talking over it. The USB MIDI port won't give you audio, just

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I was wondering if there is any way to encrypt IAX traffic? I am aware of the ability to use md5 or RSA for authentication, but I'm talking about the packets themselves, after authorization has already occured... Forgive me if this is documented somewhere, but I all I could find online was a presentation with the statement "there has been talk of adding encryption to the IAX

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio

Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix

Does anyone have a good suggestion for a automated solution to record calls on certain interfaces and easily archiving them in a way which is easily matched against CDRs? Also can someone suggest the appropriate protocol to archive the recording when the conversations are transpiring in ulaw. Basically a nice cost effective trade off between CPU and disk space for medium call load. Thanks Miles

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

Does soxmix works with asterisk ver. 0.9? I have ver. sox-12.17.5 on Gentoo but the option "m" does not combine two WAV files (In and Out) into one file. I have two separate files in /monitor folder. exten => 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 711,2,Monitor(wav,${CALLFILENAME},m) exten => 711,3,Dial(${sales_support},20,r) exten =>

The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. Now I want to record a "recitation", which consists of me playing my piano and talking over it. I have a Casio CDP-230 digital piano but I've never tried hooking it up to my computer and making a

Hi, I'm testing Asterisk installation on my Slackware Linux Box, I am making calls from kphone. I've noticed that standard sample voice files attached to * distro (*.gdm files) havn't good quality. What can be done to improve it? I'm using alsa drivers. BR, Corvin

Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together

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

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>