similar to: Audio files problem - as usual

Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets.

Hi- let me start off by saying I'm a newbie to Asterisk and this list and I'll also apologize up front for stupid questions. I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel gateway set up. I used the configurations from the O'Reilly article and I haven't even set up voice mail (the only change was to add the iaxtel entry). My problem is the audio out from my

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

On 7 Nov 2017 13:36, Lucas Clemente Vella <lvella at gmail.com> wrote: 2017-11-07 11:10 GMT-02:00 encrupted anonymous <sergeinakamoto at gmail.com<mailto:sergeinakamoto at gmail.com>>: did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable

I have been using X-Lite on FWD without any troubles and recently became interested in trying asterisk. I am able to register from X-Lite and dial a number - I've tried dialing some of the sample numbers in the sample extentions.conf file, like 500 and 1234, they appear to dial correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some

Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call

For some reason the attachment did not go through. Re-attaching. From: Mukund Raman Sent: Wednesday, April 01, 2015 6:12 PM To: 'opus at xiph.org' Subject: Opus multi-stream/surround: Audio corruption on decoded content Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Hi, I've got two SIP clients, one is X-Lite on NT, the other is KPhone on Linux and when I try either the echo test or voicemail demos, they fail to send any audio. They are both set up as type of "friend" in sip.conf so that they can send and receive calls. Using an IAX client like Gnophone, I have no problems. The troubling thing is that I'm almost certain that this was

Greetings! I have made great progress thanks to this group. My Asterisk seems to be working for the most part. I am using the following equipment/software: * HP Vectra VL - Pentium Pro CPU - 256MB RAM * Redhat Linux 8 - Loaded straight from distro CDs as Developer Workstation - latest updates from RHN * Asterisk (latest as of two weeks ago when I used CVS checkout) * X-Lite SIP Client on a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all. I've been looking for any other Free (speech) codecs and haven't found any. So it seems that it's rather important that Ogg Vorbis stick around and be at least a little "successful" (Whatever that means, I guess not getting sued would count? ;) I have no idea what kind of maths is involved in creating a new Free

Is there support for encoding/compressing 8KHz sampled speech? Am I heading in the right direction with Vorbis? I have heard results with 44KHz audio and am quite impressed. As always, any help appreciated. Regards, Paul -- Paul McHale Work: 937-320-5495 Double E Solutions Mobile: 937-371-2828 1435 Edenwood Dr Fax: 413-215-3232 Beavercreek, Ohio

I bought a ADSP-BF533-EZkit-LITE V2.1 to develop a speech codec application using SPEEX. I have tested SpeexEcho project from both BlackfinSDK-Rel201 and BlackfinSDK-Rel310. When disabling SPEEX processing, i.e. using PASS-THROUGH mode, the output voice is ok, which is exactly as the input voice. However, when enabling SPEEX processing, the output is bad, no matter how to set the parameters in