similar to: Recorded file processing app wanted

> 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

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to

I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If yes, then he would record a temp file with the additional message and when done, I want SOX to

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

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

Hi, I've been trying to get MoH files to sound decent. I've got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds really bad. I don't expect concert hall quality of course, 8000KHz being what it is, but is there a better way to convert

On Mon, 18 Jan 2016 16:09:17 -0200 "Ethy H. Brito" <ethy.brito at inexo.com.br> wrote: > On Mon, 18 Jan 2016 09:38:52 -0800 (PST) > Steve Edwards <asterisk.org at sedwards.com> wrote: > > > On Mon, 18 Jan 2016, Ethy H. Brito wrote: > > > > >> how to flush user input before READ()? > > > > How about a read() to a dummy variable

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there

Hi all, I have to play a alaw file with .wav ext. How can I do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/de46328f/attachment.htm

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 all, A long time ago I built an Asterisk system that plays IceCast streams via moh. extensions.conf: Exten => moh,1,Set(SIP_CODEC=ulaw) Exten => moh,2,Answer Exten => moh,3,MusicONHold(test_new) Exten => moh,4,Hangup musiconhold.conf ; test_new [test_new] mode=custom application=/etc/mystreams/test_new.sh test_new.sh #!/bin/bash wget -q -T 120 -O -

I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303

Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17]

Steve, These are live streams of events so I can't simply rip the audio. As I mentioned at the end of my email putting in a sleep did help a bit however there are only so many streams Asterisk will grab nicely at once with out spiking the CPU. I also tinkered a bit with real time here is what I found: 1) If we have cachertclasses=no then Asterisk will only pull the stream if some one is

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".

Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select

On Mon, 6 Feb 2017, Tech Support wrote: > We were able to develop a feature to send the call to voicemail about > 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) delete the message without listening to > it, or (3) listen to the message when it was most convenient for them. > That way, they were in control and things were

Hi: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated. ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100202/ee00a520/attachment.htm