similar to: asterisk and gnokii on same server: scratchy sound

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following 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

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 all, sorry for the OOT. I've been trying to setup a sms gateway for a couple of sleepless nights now. I use Itegno 3000 USB gsm modem, gnokii sms manager, and playsms a web interface for sms http://freshmeat.net/projects/playsms/ on FC4. Here's what happen: 1. Using minicom, I can send sms 2. Using xgnokii (x interface of gnokii), I can send sms 3. Using playsms, I CANNOT send sms, it

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

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, I'm trying to package gnokii (using FC6 SRPM as base). I can build it without problems, my problem is that I want to upgrade it from CVS version. I download the cvs version (from cvs, because "current" tar.gz appears to be broken) but ... ?How can I convert this CVS version to a "peseudo release" to make a diff from 0.6.14 and add the patch into the SOURCES? Thanks

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

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

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

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

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Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network

Hi I want to setup a SMS system for Nagios on a 5.9 box. I read in a blog that the two modems Huawei [ E160G | E176 ] work with 6.3. Anybody any experience with those modems and do they work with 5.9? Are there any other devices that are better/recommended? Thanks Jobst -- 186,262 miles/second : Not just a good idea, it's the LAW. | |0| | Jobst Schmalenbach, jobst at

----- Original Message ----- From: Gene Cumm <gene.cumm at gmail.com> > Any chance you could sent that as a pcap file Will do asap. Thanks Vieri

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

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

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