similar to: TDM 400 hardware(?) issue

On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport <malcolmd at sangoma.com> wrote: > Howdy, > > There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz. > > Try that. > I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work correctly. For one

Hi I wonder if anyone has any sugestions I have had a TDM400 for a couple of years, and I have always had problems with noise on the line, so tonight I have been doing some research and have found that if I load the CPU dahdi_test has almost perfect results and no noise dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.997% 99.999% 99.998% 99.997% 99.999% 99.998% 99.998%

Seems like I post about this about once a year, when it's time to upgrade Fedora. I first got this error trying to compile a patched version of dahdi-linux-2.11.1; I noticed that there is now a dahdi-linux-complete-3.0.0+3.0.0, so I tried that one with the same result. If I compile it while running kernel-4.16.8-300.fc28.x86_64, it compiles fine, but when I try to compile it while running

I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test !!!!!!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!!!!!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!!!!!! COMMON_MIN_TRES iREG 1E = 0 should

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [s at macro-dialout-trunk:19] Dial("DAHDI/1-1", "DAHDI/g0/9239220,300,") in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the

I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an

Has anyone figured out how to power a Digium TDM 400P card in a Dell 1750 server? I opened the server and noticed that there is no access to 4 pin power to power the card. Is there some sort of adapter that I need to power the Digium card in a Dell Server? I see that the 1750 is listed on the Wiki. How have others powered the TDM400P in a Dell 1750?

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Hello all, Recently I bought a TDM02B digium card to conect to the PSTN. We pluged it on a Pentium IV 2,8 Ghz, Asus Motherboard, but when we try to start asterisk, the box hangs. Someone have the same card running with asterisk in a similar machine? Could you tell me your box hardware details? Thanks for your time, Ismael Gil.

Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten => fax,1,Dial(DAHDI/1) ; the fax machine exten =>

I'm running a TDM-400P with 2 x FXS and 2 x FXO. I'm finding that there seems to be an odd relationship to sound quality on the card to my local when connecting via a SIP client. When I'm on my local network, if I connect to Asterisk via a SIP client (such as x-pro), and dial an outside line through the card, sound quality seems quite good. However, when I'm at a remote

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: Thursday, December 14, 2006 4:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Hardware TDM Switching > > asterisk@nicox.org wrote: >

Hi Everyone, Can you tell me how many concurrent TDM (Dahdi) calls that a single asterisk box can handle. Configuration is as follow : Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9 Also do you know a good tool to stress out asterisk? Kind regards -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* --------------

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get: