similar to: Just static on TDM400P (not even a dialtone)

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like. I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my expectations too high? Is it that once I'm using Asterisk as a gateway to the PSTN, I

Hi, I'm relatively new to Asterisk, and I'm looking to build a tiny system for home use. For context, at home I've got a line from Vonage (last I heard, they won't give out SIP credentials and let you use random hardware/software), which comes out as an analog line with a dialtone, and I'll be treating that as if it were a regular PSTN connection. I also work from home,

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B. But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 0.1B. Rarely, the phone will crack out an occasional weak and abrupt beap, but never a normal ring. Otherwise Asterisk and TDM400P

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

I think I've configured everything to have stuttered dialtone on my analog phones (it works fine for my SIP phones). But I still don't have it:-( I'm using asterisk 0.9.0 and zaptel 0.9.1. In voicemail.conf: [local] 21111 => 987654,Robert Withrow,email@here.com In zapata.conf [channels] ... mailbox=21111@local channel => 1

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

I just got a TDM400P with 2 FXO cards. I got it all configured and I can place and receive calls. I seem to be getting static on the call, mainly when I speak. E.g, if I call someone, I can hear them just fine, but they would hear static. Not a lot...more like a constant background "hissing" noise. What can I look at to try and troubleshoot what is going on, or is this normal? I

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

G'day list, It appears I got ahead of myself in believing I'd solved my CND (caller id) problem. My callerid.c change got it working on one of the handsets I have, but not the other. (Recap: The AU CND standard has an increased delay between the end of the ring signal and the start of the CND FSK data -- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030). I've

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear but it sounds like you are underwater, there is an annoying hiss or buzz on the line as well. If I hang up and pick up another house phone

I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from