similar to: scratchy sound

Hi, I have an audio quality problem regarding IAX2. 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

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

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

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

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

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

Dear All, I have move 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

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 all We 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

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

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

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

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

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

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi, I have two types of UPS devices and one of them (APC) gave me a false "low battery" alert. I can't reproduce the problem... Also, there was nothing apparently wrong with the UPS at the time of the false alert. All I could do was grab the driver values within a minute after the "low battery" notification. These are the values: battery.runtime.low : 120