similar to: FXS ports on TDM410P card...

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, I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill from the phone company. I'm investigating, but can anyone provide some feedback on what's happened here? I'm investigating how this happened as well as what types of arrangements can be made with the

Hello, I have just installed an Asterisk server with a Digium TDM410P card with 3 FXO modules (no module in the 4th slot). It's lived on two different machines (a test machine, which had Linux kernel 2.6.28, and a new dedicated machine which has Linux kernel 2.6.32). On the test machine (2.6.28), I used the Zaptel drivers. Once the kernel modules were loaded, the lights on the TDM410P came

Hi! I have a really weird problem with some of my servers, namely all the ones running Centos5 When I try to download a file from the server to a machine outside our Cisco 6500 router/firewall, the download hangs about half the times (15 out of 40) when less than half a megabyte into the transfer (varied from 76 kb to 496 kb). One server has a portchannel (Cisco speak for ethernet bundle),

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver? Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

As recommened I got the new firmware for my echo cancellers and it solved hte problem with the agressive echo cancelling causing half duplex audio. I have to say, so far these cards are far superior to the previous models. The sound quality is hugely improved (enough to really notice which is alot) and the echo canceller works way better than the software ones. My system seems to like these

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed

Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: > Le 28/01/2015 22:03, Steven McCann a ?crit : >> Hello, > > Hi > >> >> I'm investigating a situation where there was a hundreds of minutes >> of >> calls from an internal SIP extension to an 855 number in Cambodia, >> resulting in a crazy ($25,000+) bill from the phone company. I'm

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

You don't mention if the phone is remote, or local. Although you do mention it had a default user/pass. If the UI of the phone was/is accessible from the I'net, the GUI does have the ability to place a call from it, that is one way the calls could have been placed. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

A previous issue has popped up and once again I'm out of ideas. During the evenings it seems that the TDM channels will spike (dahdi_monitor) and will refuse to listen for audio of any type, this includes DTMF. The only resolution I know of is to stop Asterisk and restart the dahdi service, but that's not a solution. All channels look like this, even the FXS. [root at asterisk Hardware]#

On Sun, Oct 16, 2016 at 11:39 PM, C Bergström <cbergstrom at pathscale.com> wrote: > Is there any particular reason you're not using the open source NVIDIA > Tegra drivers? > > This is what is being used for the TX1 (Sorry I don't know if it's the > same for the TK1) > http://nv-tegra.nvidia.com/gitweb/?p=linux-3.10.git;a=summary That's ... not particularly