4.1.1. How to install and configure Wildcard TDM400p
The wildcard TDM400P is a half-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC.
In this tutorial we will give you an example which includes one FXO and one FXS station interfaces. TDM400P card with such modules is called TDM11B. Below you will find description of the name model used by Digium.
The Wildcard TDM11B is a half-length PCI card that supports FXO and FXS interfaces for connecting analogue or ADSI telephones and regular POTS through a computer. TDM11B card is from the TDM400p family. This card can be extended with additional FXO or FXS modules (maximum 4 modules per card).
Can we find what modules are plugged in the card just from the name? Yes, the name is divided in several parts: TDM X Y B. X – shows how many FXS modules are in the card, Y – is the number FXO modules. So TDM11B – has one FXS module, and one FXO module
1.PREREQUISITES
To install such Wildcard we assume that you already have installed Asterisk PBX on the PC you would like to install the TDM card.
For the hardware requirements you need minimum 500Mhz Pentium III or better with 64 MB.
2.INSTALLATION
Find empty PCI slot
Have a look at the card before placing it in the PCI slot. Find empty PCI slot and plug the card there, next you have to plug power cable in the card. There is a red and a green module on it:
Green Module is FXS – Foreign Exchange Station
Red Module is FXO – Foreign Exchange Office
An FXS device initiates and sends signals to an FXO device. The telephone that receives the calls is the last FXO device (if you have several FXO devices) and when the signal is received from the FXS device the telephone has to ring.
Plug the cables
To connect the device to your network – connect the FXO port to the outside lines and local PSTN phone to the FXS port. The FXO uses FXS signalling and the FXS device uses FXO signalling.
When you have everything connected, turn on the Asterisk PBX PC and configure the device there.
Did your OS recognize the card
By typing ‘lspci’ you will receive a list of all the PCI devices you have. Note that Digium cards are recognized under different name:
- Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) for the TE410p / TE405p
- Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface / Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface for the TE110p or the TDM400p
- Network controller: Jens Scoenfeld for the TE110p or the TDM400p
Zaptel installation
If you have not install the zaptel on your asterisk you have to do it now. The PSTN calls on asterisk are passing through Zap channel. For zaptel you need to install the zlib1g-dev module.
apt-get install zlib1g-dev
Now in order to compile it
cd /usr/src/asterisk/zaptel
make linux26
make install
And then recompile the Asterisk
cd /usr/src/asterisk/asterisk
make
make install
3.CONFIGURATION
In order to make calls you have to register users and extensions (respectively in iax.conf and extensions.conf) and to adjust the zaptel protocol settings in /etc/zaptel.conf and /etc/asterisk/zapata.conf.
configuring zaptel.conf
In zaptel.conf it is enough to set the type of signalling for FXO and FXS. As we mentioned above FXO uses fxs signalling and FXS uses fxo signalling. For FXO devices you can use one of the following signalling methods.
fxsls : Channel(s) are signalled using FXS Loopstart protocol
fxsgs : Channel(s) are signalled using FXS Groundstart protocol
fxsks : Channel(s) are signalled using FXS Koolstart protocol
And for FXS one of the types standing below
fxols : Channel(s) are signalled using FXO Loopstart protocol
fxogs : Channel(s) are signalled using FXO Groundstart protocol
fxoks : Channel(s) are signalled using FXO Koolstart protocol
Here is my simple configuration
Note that fxoks=1 means that the FXO Koolstart protocol signalling used for the first device which has to be FXS. So have in mind the order of the devices on the TDM card when you register the signalling for each one.
With loadzone and defaultzone you can define the tone zone which to be loaded where the zone is a two-letter country code.
configuring zapata.conf
In zapata.conf some channel configurations are to be made. As I have just two devices, an FXS and an FXO, so I will have just two channels. Here is how I configured my zapata.conf.
As you see there some ‘global’ settings above – they take effect for each channel defined.
Context is the extensions context where the channels will be used (if omitted the default context will be used) Usecalledid is for using or not called id (may be omitted) Hidecalledid is whether or not to hide the caller id (may be omitted) Immediate specifies whether a channel should be answered immediately or not
Signaling is the signalling defined in /etc/zaptel.conf. Just add an underscore before the protocol type (signaling = fxo_ks) Echocancelation is used for enabling echo cancellation. The valid values are: ‘yes’, ‘no’ or a power of two from 32 to 256. Group is used for device logically the outgoing calls. Groups can be defined from 0 to 63 as well as multiple groups can be specified. Channel is channel number for each channel.
User registration
I also have a user registered in iax.conf, but you can also place calls without any user registered from CLI (see the dial command).
Register extensions
And the last step is to register valid numbers in extensions.conf.
Here when you enter the standard extension an operator picks up, the variables concerning the caller is displayed in the CLI – the name of the caller, the number of the caller and the id of the caller, then some welcome message is played. After this you can leave a mail on the voicemail and then the operator hangs you up. You can register a voicemail in /etc/asterisk/voicemail.conf by writing the example below. Here you can learn more about voicemail.
[vm-test]
voicemail context is created
1111 => 123,gogh,gogh@some_domain.com
1111 mailbox is created with password 123 and assigned to user named gogh with email gogh@some_domain.com to notify the user when new message is received.
When you dial some extension like 0ZXXXXX (where ‘Z’ is number from 1-9 and ‘X’ is number from 0-9) you are send to a Macro which places a zero in front of the number and dials the new number. The reason to have additional zero in the beginning of the number is that all outgoing calls for my server must have a zero in front.
exten => s,2,Dial(Zap/2/0{$ARG})
this dials the number you want through Zap on channel 2. As you saw above in my case channel 2 is FXO channel with fxsks signalling and this channel is for outgoing calls.
I have also created a conference room. By dialling 9090 you can enter in it. Conference room can be defined in /etc/asterisk/meetme.conf. Here is a simple example:
[rooms]
conf => 9090[,pin][,adminpin]
This creates conference room with number 9090, without authorization when entering.
By dialling 1001 you call to the user we registered on asterisk PC.
Extensions 1111 and 2222 dial some mobile phones – in the first case through group 2 Dial(Zap/g2/…) and in the second through channel 2 Dial(Zap/2). In the example here channel 2 and group 2 are the same and they use the FXO channel, i.e. outgoing calls.
Extension 3333 dials channel one on Zap.
exten => 3333,1,Dial(Zap/1)
This dials channel 1 which is controlled by FXS. As we said in the beginning the local analogue line is connected to this place, so your local PSTN phone will be dialled.
4.TECHNICAL DATA*
Target Application:
- Small Office Home Office (SOHO) applications
- Gateway termination to analogue telephones
- Add inexpensive analogue phones to existing PBX
- Wireless point-to-point applications between Asterisk servers
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Praveen (3pbwighccp at yahoo dot com) 25 September 2015 12:44:49 HelloI just bought an IBM Netvista 2800 8364-EXX off eBay to do the exact same thing: Build a small, quiet Asterisk sevrer with a TDM card to handle a PSTN line.At this point, it complains that no mouse/keyboard is plugged in, although they are, but besides that, I have a couple of questions for you:- would you recommend buying the HD25-I or is a CF card good enough?- is installing Linux and Asterisk on this thing doable by a semi-experience Linux user, or is it a bit more involved?Thank youGilles.
franklet (frank_semilla at yahoo dot com) 12 May 2009 08:21:25 Hi Good day..
Can you teach me how to solve the "2 channels to configure" issue? Ive done all the steps but still I did not resolve it. I am using TDM400P just like above. And why is it that theres no documentation with regards to Zap configuration?I always find how to configure Iax and Sip in the internet.. =( how about Zap? please help me with this...thanks
Nikhil (chava dot nikhil at gmail dot com) 26 August 2008 09:27:26 can anyone help I had a problem
I am using TDM2400P card and I sucessfully installed Asterisk when i dial a number it is showing that call established and no response after that can any one say what might be the problem
when I type #ztcfg -vv
it shows like this
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Slaves: 16)
Channel 17: FXS Kewlstart (Default) (Slaves: 17)
Channel 18: FXS Kewlstart (Default) (Slaves: 18)
Channel 19: FXS Kewlstart (Default) (Slaves: 19)
Channel 20: FXS Kewlstart (Default) (Slaves: 20)
Channel 21: FXS Kewlstart (Default) (Slaves: 21)
Channel 22: FXS Kewlstart (Default) (Slaves: 22)
Channel 23: FXS Kewlstart (Default) (Slaves: 23)
Channel 24: FXS Kewlstart (Default) (Slaves: 24)
fxsks=1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24
load zone = uk
default zone = uk
please help me thanks in advance
tom brown (nospam at nospam dot com) 21 March 2008 02:37:41 Don't copy the mistake out of the oreilly book (page 78) and mispell signalling ... I did. That caused the chan_zap module to fail to load. One sign of that is that "core show channeltypes" will not list a Zap type. If that is the case you can try running "module reload chan_zap.so" and it may give you some diagnostics as to why the module is not loading.
-Tom
souvik (sadhu dot souovik at gmail dot com) 04 March 2008 10:42:52 I am not getting any dial tone in my desk phone which is connected to FXS port.
I am using TDM11B card.
everything looks fine.
phone is getting power but no dial tone.
please suggest what is wrong in this!!
Thanks in advance
souvik
Ashwini (ashwini dot pict at gmail dot com) 30 January 2008 13:26:15 Connecting two asterisk servers in same network
We have two asterisk servers in a same LAN , we can able to make calls using both the asterisk servers individually . But we need to communicate between two asterisk servers and to make call to the number exist in the another server. i.e. if Server A has number 5060 and server B has number 8891 , i have to make call from 5060 to 8891 .
I have searched online for several times and tried the options given , but nothing seems to be working. Please guide me in this issue
Marcelo (marcelodanza at gmail dot com) 04 January 2008 20:22:56 I would like to recive tutorials
Jason (jason at 3-link dot com dot tw) 13 November 2007 06:52:50 I use TDM844P Card.
When I dial the local number through the PSTN,it can't work.
The message shows:
[Nov 13 11:34:52] WARNING[19942] channel.c: No channel type registered for 'zap'
[Nov 13 11:34:52] WARNING[19942] app_dial.c: Unable to create channel of type 'zap' (cause 66 - Channel not implemented)
[Nov 13 11:34:52] NOTICE[19942] cdr.c: CDR on channel 'SIP/09015126502-081bc330' not posted
Can someone help me why I get this message?
Thanks for share!!
conrad (conrepos at gmail dot com) 11 September 2007 11:26:31 this is insanity , the technical support team cant even reply to the problem post above , have same problem of you guys but didnt post it
hungdm (dmhung1c at yahoo dot com) 09 July 2007 12:12:54 I use TDM40B
I configured as flow:
RUN asterisk -vvvvr is OK
ztcfg -vv is OK
BUT Why my phone (conected to card, I try to connect to all port) Can not received Tone when i pick up handset. Whyyyyyyyyyyyyyyyyy??
Please Help me!!!!!!!!!!!!!!!!!!!!!!
saleem (greenday3600 at hotmail dot com) 12 June 2007 16:00:08 please can any body provide me of circuit to wildcard TDM400P
i want to create one
zoa (support at asteriskguru dot com) 07 November 2006 17:54:57 Do you actually have any zaptel kernel modules loaded ?
Paul Velasquez (paul dot velasquez at ictec dot com dot bo) 07 November 2006 16:45:00 I have configurated zaptel.conf, zapata.conf, and extensions.conf, and when I try to make this call, I've got this maessage. Please Help me.
[Nov 7 11:25:36] WARNING[9791]: channel.c:2736 ast_request: No channel type registered for 'Zap'
[Nov 7 11:25:36] WARNING[9791]: app_dial.c:1077 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
Daniel (danboh at gmail dot com) 30 October 2006 16:06:59 hi all!
I need to receive more than one call simultaneously through a POT line. In freePBX I configured a ZAP trunk, and in Maximum Channels I entered 10. However, it doesn't seem to work.
Any ideas?
Thanks in advance.
javier (jgonzalez at conatel dot com dot uy) 13 September 2006 18:50:23 I use usercallerid=yesand callerid="Euroset 805" <1001> on my FXS modules of TDM400P but on the softphones always a call from these telephones is identified as unknown.
Where should I look to correct this issue ?
Thanks in advance
Elmer (elmer at diavox dot net) 06 September 2006 15:40:08 Hello everyone. I have a TDM400P card. Is there anyone here that could help me configure it. Or if there is a step by step manual for it please send me the link.
Thanks.
Sebastian Kisiel (kisiel at schwedler dot com) 29 August 2006 19:40:31 I was not able to load the wctdm module correctly.
Error messages was:
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
The TDM400P device was bound by a module called netjetpci before the wctdm module could do this. So, avoid loading the module netjetpci before wctdm!
I hope this could help anyone. ;-)
madman2 (robert_campbell at btconnect dot com) 04 August 2006 17:47:20 echo cancellation seems to be a bit of a black art with this card.
What's the best strategy to test/reload/test settings.
Any thoughts on whether or not this card can be used with the analogue port on a UK ADSL connection successfully???
Tolkien (aliarla02 at yahoo dot es) 24 July 2006 19:53:22 I'm having problems with Caller-ID do you have the configuration for spainsh PSTN network , thanks.
Muhammad Ali Mansoor (malimansoor at hotmail dot com) 05 July 2006 15:27:55 I have TDM11B card in my Asterisk server. I can manage incomming calls but I am unable to configure my FXS port for outgoing ones. My Asterisk CLI is unable to execute because of some errors in chan_zap.c It also gives some warnings like unable to configure Channel 1 (My FXS channel) There is no such device.
Can somebody help me out ?
Stephen Gleeson (sg at cisa dot asn dot au) 13 April 2006 01:52:53 I am having some issues with this card, if call is not closed down for some unknown reason correctly. The line stays engaged, and other than a reboot of the system I have not been able to find a way to clear this. If undetected after approx 24 hours, all lines on the card become unusable.
Steve (sblurton at yahoo dot com) 08 April 2006 01:45:46 I have a TDM04b I need to set a pause before dialing due to slow QWEST pots LINE
vinit (vinitsid at gmail dot com) 17 February 2006 11:36:48 we r having one X100P attached and configured Asterisk and zaptel properly. Channel has been established by command ztfg command and it shows that one channel is established.
when v dial through asterisk it shows error:
WARNING[3614]: channel.c:2535 ast_request: No channel type registered for 'Zap'
NOTICE[3614]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)
please help me why i cannot call to PSTN lines.
Miguel (mnaveda at gmail dot com) 04 October 2005 22:41:44 Hello, i have a trouble wiht a TDM22B, in asterisk console show me follow message " Unable to create channel of type zap" any one can help me. i have this configuration:
Zapata
[channels]
context=default
signalling=fxo_ks
echocancel=yes
echotraining=400
group=1
channel=>1-2
Gobinda (gobinda at bigmastech dot com) 10 September 2005 10:36:08 Where i get the complete g729 without lincence key.
Robert Murray (robertm at marco-na dot com) 13 August 2005 23:49:05 Ali make sure the context you have in your zapata.conf for the channels has a context in your extenstions.conf if not the incoming call might never get picked up or other problems depending on you extenstions.conf
If you notice in the above examples the context is set to test.
Then in the extenstions.conf file he has a test context. The s extenstions are a starting point for the incoming calls.
Robert
ali (ali dot kia at caramail dot com) 21 July 2005 12:59:47 hi all
i tried to manage incoming call using TDM04B but i don't know why it doesn't work,i can manage succesfully the outgoing ones
could any body help me