similar to: How to create distortion, echo, and chopping sound in a SIP trunk?

hi All, Our configuration is, ISDN PRI lines connected to Asterisk Server, SIP users connected to Asterisk We route calls from ISDN PRI to SIP users, We did a CVS upgrade few days ago, now the sip users(CISCO phones) are experiencing, distortion of voice while they are engaged in a call. ie, SIP users can here the call clearly, but the outside caller heres distorted voice. ie, to

For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel

If I encode something in ultra-wideband, can I decode it in wideband by chopping off bytes in every frame? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed.

Hello, I have been facing an issue that voice is getting distorted sometimes in MOH (MusicOnHold) application. I have checked and confirmed that lame is properly installed, even tried native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH can't be eliminated. I came to know about requirement of timing device for MOH and MeetMe and a very good illustration by Andrew

Full_Name: Thomas Zumbrunn Version: 2.6.1 OS: GNU/Linux (openSuse 10.3 2.6.22.13-0.3-default) Submission from: (NULL) (131.152.125.199) With my particular X11 settings, the width to height ratio of the x11 device is distorted, with the width being half that of the height. This results in wrongly proportioned plots, wrongly positioned text etc. Other devices are not affected. I already asked

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

Hi good people, Just to inform: I finally fed FFMpeg2Theora with a .VOB input, and the .OGV output produced played back beautifully in VLC, both audio and video. However, the same output .OGV file gave clicks and pops in the audio when fed through Directshow programs. I played it through Windows Mediaplayer as well as through Winamp, with the same crackles in the audio, and this leads me to

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

Hi all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This

If I have a two column data frame like: > dat <- cbind("x"=c(1:100),"y"=c(100:1)) How can I create an array that splits every ten rows of that data frame into a third dimension of an array so that: > newarray[,,1] ,,1 x y 1 100 2 99 3 98 ... ... 10 91 ,,2 x y 11 90 12 89 ... ... ... Thanks. [[alternative HTML version deleted]]

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except

We are experiencing slightly distorted audio with playing of recordings on our Asterisk server when the call comes in over our Eicon Diva Server BRI card. An example is an incoming call to IVR and playing some of the standard Asterisk voice prompts. Note that there is no audio problem with internal access to the same recording. Neither is there a problem with calls not involving the playing of

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

I'm new to * and I have my * box connected to another * box located at my ISP via iax2. Using ilbc everyting works fine but with any other codec I've tried (gsm, ulaw, alaw) there is severe distortion on the sound going out of my box as well as a second or two of delay. The incoming sound quality is fine. This happens even if the outgoing sound is coming from the voice mail prompts on

On Tue, Feb 18, 2003 at 06:09:43PM -0500, Jean-Marc Valin wrote: > Le mar 18/02/2003 ? 17:38, John Hayes a ?crit : > > If I encode something in ultra-wideband, can I decode it in wideband by > > chopping off bytes in every frame? > > All you have to do is use the --force-wb switch with speexdec. It will > decode as if the file were wideband, ignoring the ultra-wideband

Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote

Ok maybe it's just my imagination, but it looks like the one-line "duh" fix I committed from Andrej Vakrcka <ander@cauldron.sk> this week DRAMATICALLY reduces distortion in encoded files. Take a look and compair. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. I did a little test. This sounds fine... exten => 1000,1,Answer exten => 1000,n,Wait,1 exten =>