search for: scratchy

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of getting someone with a clue are slim to none....

...from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line. I'm experiencing two issues; 1) There is a latency of .5 - .8 seconds between me and the USA. 2) I have been in two calls where my voice has been describes as 'Scratchy'? I'm using a SIP Phone from SJ Phone, and a Plantronics USB Headset. In my Asterisk box I'm using the Quad T1 card. Any tips on how I could get around these two issues? I can understand the latency issue, what is contributing to the 'Scratchy' sound? I have not had this issu...

I'm currently working out the config bugs on my * box and I'm noticing that the meetme is very scratchy. As in not usable scratchy tho I can hear the audio it sounds like when you talk through a fan. Anyone have any ideas? Linux 2.6 with RTC installed. Using stable release and SIP devices. -Don

Hi, Has anyone tried to use gnokii to send/receive SMS messages via serial or USB with AT commands while running Asterisk? Some of my calls have a "scratchy sound" once in a while. It doesn't seem to be due to packet loss but some kind of interference (CPU is ok, etc.). I've noticed some coincidence in time between this scratchy sound and the gnokii process. I have a bash script that calls gnokii periodically to send/receive messages. The...

...sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can't determine the possible cause of this kind of distortion. Maybe an expert ear can give me a clue. It shouldn't be the Asterisk server's CPU usage. Could it be a network issue between the Asterisk system and the SIP client? (it happens with SIP ha...

...run an * 1.2.4 under FreeBSD with ztdummy kernel module. zttest reports 99.9something % of accuracy, so timing should be fine. SIP connections work fine, but we have a strange problem with IAX2 connections. When an IAX2 call originates from the FreeBSD Asterisk to another Asterisk, the sound is scratchy (sounds a bit like a 50Hz ground loop). It's not a problem of the 'other' asterisk, as the problem could be reproduced with * 1.2.5/Linux and * 1.4.2/Linux. If the IAX2 call originates from another * to the FreeBSD one, the sound is clear. Format used is alaw. (ulaw also shows that...

I was wondering if someone could help me out. I have my * box configured to give the caller a menu (press 1 for sales, etc.) the only inbound connections to * are via SIP (although a X100P and TDM400P are installed, just not physically connected to any phone lines) once in a while the playback will become unrecognizable to the caller. All of the menu prompts were recorded in GSM format using the *

On Mon, 2005-01-10 at 08:01 -0600, asterisk-users- request@lists.digium.com wrote: > > I am new to asterisk but learn a lot about it to this mailing list > and > > wiki currently i am facing problem about sip phone i have "PA 1688" > > chipset ip-phone and i have iptel.org sip account i registered > locally > > and through iptel.org comfortably my problem

I have a number of servers with TE405P cards. The servers are DELL 1850's (which I _NOW_ see are listed on the digium "not recommended page" because of the ethernet interface). The problem I have is only during bridged calls. If I place a call into a service hosted on the box, or out to a VOIP phone, audio is crystal clear. If place a call "through" the box (a bridged

Hello Group, I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have "PA 1688" chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy

Hi, it took me 2 days to get my asterisk box running, so now I completed and I am disappointed of the sound quality. When I call other people their voices sound somewhat scratchy. First I thought it might be a codec problem, but I also recognized it during the ring tone or even the DISA connect tone. Sometimes it is better quality and sometimes more scratchy. Where might be the problem? I am using asterisk with bristuff and a HFC card on Debian 3.0. Perhaps someone kno...

...s the following debug info: > > Initialising global parameters > params.c:pm_process() - Processing configuration file > "/usr/local/samba/lib/smb.conf" > Processing section "[global]" > doing parameter workgroup = SIMPSONS > doing parameter server string = Scratchy Test > doing parameter netbios name = scratchy > handle_netbios_name: set global_myname to: SCRATCHY > doing parameter printcap name = /etc/printcap > doing parameter printing = lprng > doing parameter log file = /var/log/samba/%m.log > doing parameter max log size = 0 > doing...

...ator%myadminpasswordhere provides the following debug info: Initialising global parameters params.c:pm_process() - Processing configuration file "/usr/local/samba/lib/smb.conf" Processing section "[global]" doing parameter workgroup = SIMPSONS doing parameter server string = Scratchy Test doing parameter netbios name = scratchy handle_netbios_name: set global_myname to: SCRATCHY doing parameter printcap name = /etc/printcap doing parameter printing = lprng doing parameter log file = /var/log/samba/%m.log doing parameter max log size = 0 doing parameter security = domain doing p...

It appears that both the echo canceller and the noise reducer (NR) introduce rather severe clicking artifacts when presented with audio that has slightly overloaded the A/D converters. I am talking about speech that sounds just slightly distorted, due to clipping, when simply played back. If I pass that speech through the echo canceller or the noise reducer, it acquires really loud clicks and/or

...AX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of interference). So lately I setup calls to go through the SIP trunk and audio quality is OK (no "third overlapping noise"). This is happening between Asterisk 1.4.31 and a 1.2.40. I'm wondering if there's something I can tweak in IAX2 to el...

...terface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO CONNECTIONS WHEN USING THE FXO PORT. Incoming calls were erratic, outbound calls were almost unuseable, dialled digits were almost unrecognizable. Basically after some difficult troubleshooting the fix was: before disconnecting cable kill asterisk process remove WCTDM module remove WCFXO mo...

Hi, sometimes a ms-office file (ms word 2k, ms excel 2k) will be destroyed, when the client writes it to the server. The strange thing is, that samba writes parts of its log-files at the top and bottom of the file. Here are the first 2 lines of the destroyed file: [2002/09/04 17:03:05, 2] smbd/open.c:open_file(216) tb opened file xxx/xx/xx/xxxx/xx xx/filename_TB.doc read=Yes write=Yes

...he irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card, the problem follows the module, not the physical port on the TDM400P. Any thoughts? Was planning on deploying this server tomorrow, but now I am worried about the other three FXO ports crapping out. Has anyone else...

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

...pparently didn't do anything else), didn't work. After looking at the source I had also tried to increase the buffer size from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, and I still had the problem... I like speex and would like to use it (as I find ilbc a bit too scratchy) I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 libraries on Gentoo Linux. Can anybody help me further on how to resolve this problem ? Thanks Walter