similar to: Trailing silence in voicemail messages

What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ;

Anybody care to share a script or snippet of what they use to remove leading and trailing silence from customer recorded files? I've fiddled with sox to remove the leading; reverse the file; remove the now leading; and reverse the file again, but I'm not really happy with my results. -- Thanks in advance, ------------------------------------------------------------------------- Steve

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

I modified app.c to print out the totalsilence when recording the message in the voicemail and ... dspsilence=0 dspsilence=0 dspsilence=0 it always equals 0... maxsilence=8 silencethreshold=128 My line seems to be pretty silent... (and it also doesn't detect silence...) any hints? I'm having some problems.. Could it be because it's on the same IRQ as my ethernet adapter?

It seems voicemail recordings have broken sound. It cuts out randomly throughout the recording. Has anyone had any similar experiences? I've included some snips of my voicemail.conf Cheers, Ben ----------SNIP------- [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav ; Who the e-mail notification should appear to come from

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

Hi, Speex is great! We are using it to compress hundreds of megabytes of speech for use in our application that trains people in resuscitation. The previous version of our product used Ogg Vorbis, but after switching to Speex, we achieve fantastic compression, while retaining super quality. That allows us to cram more translated versions of the software onto each CD-ROM, making everything

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

Actually, there are probably some batch programs that could do the job. It's definitely not a job for speexenc, which I'd like to keep simple. Jean-Marc Le ven 09/01/2004 à 17:37, Asger Kunuk Alstrup a écrit : > Hi, > > Speex is great! > > We are using it to compress hundreds of megabytes of speech for use in our > application that trains people in

Hi, Jean-Marc Valin wrote: > Actually, there are probably some batch programs that could do the > job. It's definitely not a job for speexenc, which I'd like to keep > simple. Fair enough. Which batch programs should I be looking for? I had a look around, and could not find any... The reason speexenc seems like a good place to do this, is that it already have the routines to

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hello all, I am having a serious problem installing my * with a TDM04B. I made everything work, call are coming in and going out including using a GSM Box in channel Zap/2-1. I did setup voicemail like this on extensions.conf: [incoming] exten => s,1,Dial(SIP/2246,20) exten => s,2,Wait,2 exten => s,3,Voicemail(u${ME}) exten => s,4,Hangup exten => s,102,Wait,2 exten =>

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

hi, I was wondering how perfect silence frames (no audio at all) in a PCM stream are encoded by the flac encoder. Suppose I have a PCM audio file with some perfect silence frames in the beginning and at the end of the audio recording and suppose I have the same audio file but without the silence frames in the beginning and at the end (so the second file is identical except for the silence frames

I need it to be able to take a stream of PCM data and encode it. Right now it works if I am recording from a source (i.e. microphone or with "what you hear") but when I try to use another API (Team Speak) to capture the PCM data it doesn't seem to be encoding correctly. (I know the PCM data is correct. I am able to use it to create a wav file.) I'm also having a problem with

With modern libvirt, when fetching the XML of a VMware guest libvirt passes us the datacenter path (dcpath). However with older libvirt we had to guess this value, or else the user had to supply it on the command line. This commit removes all the guessing code and the --dcpath parameter (which will now give an error). This requires libvirt >= 1.2.20 for virt-v2v, released Oct 2015. ---

Hi - I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now