similar to: Multiple Repeated tones with TDM02B

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Has anyone had any echo issues with the Digium TDM02B FXO card? I purchased a clone wildcard on E-Bay for $8.00 and have had horrible echo issues.... I'm assuming it is because of impedance issues? Just wondered what people's takes were on the TDM02B FXO and echo?

Hello all, I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with zapata gain values and with the Snom mic volume, but the echos does not disapperars. the phone is

Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I

Hi list, I tried to use DISA in order to get the line when I call with my mobile phone but the system doesn't recognise my DTMF tones when I call to a SIP trunk. Everything is working Ok if I use a ZAP Trunks. I tried to google to find a solution but I wasn't able to find any. Any idea? I'm using trixbox 2 with 4 SIP trunks and a Zaptel TDM 400 card. Bye

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic Quiero configurar una tarjeta x100p i usarla con asterisk, asi que descague compile e instale lo siguiente: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 Sin embargo no logro configurar la tarjeta con exito, ya probe casi? todo. Esto aparece si ejecuto lspci: 04:06.0 Communication controller: Motorola

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

Hi, I've been avoiding investigating this issue for a while; I used to revert to a previously compiled version of zaptel & a previous kernel (as at some point I think I stopped being able to compile the older zaptel against the newer kernels) and all was well. However I've now upgraded kernels again and it seems silly to hide from the problem - here goes, let's try and fix it! At

I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this:

Dear friends, As I commented some while ago in the list, occasionally when DTMF Tones are sent, they appear in RTP Payload and in Events too, producing duplicate tones being recognized. This behavior happens in Asterisk as well as in Gateways such as Cisco, for which we had the opportunity to observe the error and extensively debug it. We ended up recognizing good digits by adjusting audio gain

Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

obviously I didn't understand somethin : By requesting an m3u file, icecast server create an handler for the stream that contained the url:port information, am I correct ? <p><p><p>> > You have the hostname and port of the server misconfigured in your icecast > config file. icecast2 obviously needs those to be correct so that it can > generate a URL to send to

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might