similar to: Detecting silence

I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at

Does anyone how you might detect a period of x milliseconds of silence using AGI ? Rgds, Stuart

Hello! As I see there is dsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence? Is it possible to change silence level, so, let's say some not loud enough background noises will be recognized as silence and only loud enough human voice will be recognized as sound? Thank you!

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

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 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

Okay, you've answered part of my question, which is "What value equals silence?". I assume then that a (decoded) frame of silence would be a buffer the size of my frame (320 bytes) full of 0's. Passing this frame (a frame of all 0's) through the encoder causes it to blowup though.. In response to your answer below, I don't think I want to overwrite the decoded

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

As has been noted before on this list, the Wait() application does not listen for keystrokes from users. Many of you, like me, have looping Background(), Wait(), and Goto() application priority chains that prompt users to enter some data, and then repeat the instructions if no keys are pressed. The problem of course is if the user doesn't start pressing keys during the Background() call

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

OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem?

Hi, I'm new to this list. I'm using Speex as the base codec in a voice chat application (with DirectSound as the audio playback/capture interface). To help me debug some of my network-related issues, I would really like to insert frames of "silence" into speech. Is there a convenient function call or API call that takes in a buffer that is the size of a frame and writes a

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

I am preparing to setup a system using Cisco 7940 and 7960's I have the 7.1 SIP firmware on them. One issue I have run into is how to silence the ringer if a call comes in and you don't want to take it. Many phones have a DND button. I know the 79XX has the DND in the menu but it is to cumbersome to go into the settings then phone preferences then the DND and select yes. Is there any other

Hi, When we pass around 9K samples of only ambient noise (no voice), the encoder which is enabled FEC+DTX is detecting only some 140 frames as non-voice (returning only TOC, no frame content). We were expecting all or more to be identified as non-voice. Our idea was to check how the decoder re-generates the original ambient noise during the silence duration (when we feed NULL to decoder) when

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hi all, I noticed some odd behavior with the Opus encoder, and I was wondering if this is a known issue. First, note that this problem occurs when the Opus encoder is created with OPUS_APPLICATION_VOIP, there is no issue if the encoder is created with OPUS_APPLICATION_AUDIO. If compiled for a 32 bit architecture (i386), the encoder experiences significant slowdowns when regular audio is followed

My static checker complains that this sprintf() can overflow but really it can't. Just silence the warning by using snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter at oracle.com> --- v2: the overflow is not possible so just leave the buffer size alone and silence the warning with snprintf(). diff --git a/drivers/char/virtio_console.c b/drivers/char/virtio_console.c index