similar to: Help with IAX , voice Distortion or Breakage.

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

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

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

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

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

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

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

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, I've set up speex AEC in our application. The echo's seem to be canceling, but the captured voice is distorted somehow. It sounds to me like low and/or high frequencies are removed. This happens even if I set my playback(echo) data to only zero's. Is this "normal" for the AEC? As far as I understand, if I send 0's as playback/echo data, the resulting data should be

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

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

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

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

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

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

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

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi there, im having some problems with my asterisk box, it seems codec is the principal cause of it. Reading in some forums i found that i can get the new codec_g729 from ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i checked it but the directory new_codec_binary doesnt exist. Anybody knows where can i found it?? Thanks for your help. Carlos Andres Medina

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