similar to: Distortion of voice after cvs upgrade

Hello, I am quite new to asterisk. I managed to connect our 2 branch offices with asterisk. In one side, our linux asterisk box is connected to the leased line going to our other office and on the other side its connected to office PBX through a channel bank. This installation is running smoothly, except for one thing. If an extension is busy, i want to transfer that call to a near by one,

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

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

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, 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

Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate

BUG/SYMPTOMS: 1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN cards (Junghanns/CologneChip) severely distort certain ISDN payload. 2.Although these claims relate to the bristuff patch, the problem might not be limited to bristuff and in fact be rather asterisk/zaptel/libpri related. 3.Signal distortion is limited to the use of the CVS version of the Bristuff patch for

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

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

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

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

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

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

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 =>

I am having almost the exact same problem. I have the following setup: Debian Woody Kernel 2.4.18 CPU: P4 1.2GHz RAM: 1GB Asterisk - Latest CVS 1 TDM400P card Codec GSM I've been chasing down bandwidth issues, but have had no luck. We are still pursuing those issues. I just started configuring IAX, so I assumed it was related to my IAX configs. We just noticed this morning that SIP is

On Samstag, 14. Januar 2006 1:47 tim panton wrote: > That is weird, you would expect IAX to do better than SIP (bandwidth > wise) My point exactly. > 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. > 2) what codecs are you using. Are the codecs the same for IAX as > for sip? G.711 alaw and

Hi All, I'm relatively new to Ogg and Vorbis and I've hit a pretty puzzling issue, and I'm wondering if a veteran could help me out. I'm attempting to decode a chained live Ogg Vorbis stream. The primary issue is that the source does not exactly follow validation (I have no control of this). The stream switches from a prerecorded segment, to a live segment, back to a prerecorded

Hi, I set "alaw = 1-7" in /etc/zaptel.conf hoping to make my zap channels the same as the PSTN. This caused the levels to be about 20dB too high, as well as being distorted. I adjusted the txgain and rxgain settings in /etc/asterisk/zapata.conf to sane levels, but now I still have rather distorted zap extension -> zap extension sound quality. What am I missing here ? zap 1-4 are

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