similar to: Why Asterisk can't cope with silence suppression?

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

>> Yes, after I stop speaking, the noise slowly starts climbing again, and >> if I peek at st->agc_gain, that's slowly climbing too. I think part of >> the trouble is that the noise in here isn't uniform white noise; there's >> traffic outside the window and people walking in the hallway outside my >> door. Each little event is enough to cause the AGC

Jean-Marc Valin escreveu: > > Well, I can change the defaults and then people with different setups > (e.g. no noise and low input volume) will ask me to change it the other > way... In the end, the only fix is to improve the AGC adaptation > decision, including (but not only) the VAD. Could that be a run time variable? So it could be adjusted somehow by a user interface, for

Jean-Marc Valin wrote: >> I've had a small case with noise suppression and AGC. I have a fairly >> noisy environment here, and with the default parameters, noise >> suppression works fairly well while I talk. However, when I shut up, AGC >> starts slowly increasing the gain until it has amplified whatever noise >> is left to levels about equal to having no

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/dde500d5/attachment.htm

Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

Hi Jean Marc, I found that the AGC API and algorithm has changed in svn head trunk. Is it safe to use it? Or is it for testing purpose only? You also said that VAD in svn is broken in a previous post, is it related to the AGC change? or can we mix the old VAD algorithm with the new AGC safely? kind regards, fredo

On Mar 28, 2004, at 8:23 PM, Jean-Marc Valin wrote: >> The st->zeta pointer isn't freed in the >> speex_preprocess_state_destroy() >> function of the preprocess.c file (alloced in line 167). It's in >> Speex 1.1.4 >> by the way. > > Oops... Thanks for letting me know. I'll change that for the next > release (in the mean time, the fix is

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

Hi everyone, We are experiencing problems with Jitter and silence suppression. Is there any doc or even sample code that help solve our problem? I am new to the list and it will take me a very long time to go through the entire archive. Any help? Thank you Badi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Hello The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() function of the preprocess.c file (alloced in line 167). It's in Speex 1.1.4 by the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their