[Asterisk-Users] confusing info from Digium andasteriskdoc aboutsetup of TDM11B

[Jim Van Meggelen]

Steve,

Thank you for testing our document, and for your valuable feedback. We
are aware that there is still much work to be done, I and apologize that
we have not done a good job of making that clear.

I have answered some of your questions below:


asterisk-users-bounces at lists.digium.com wrote:
> Looks like it's still incorrect in the first blue paragraph
> of the section on
> FXO (it's fixed in the second blue paragraph).  Also, the
> last paragraph of that
> section twice still calls the channel # 2.


> Now on to my next confusion...  The section on contexts under
> dislplans mentions a context named [incoming].  This isn't a
> context that's mentioned anywhere
> before this and it's not at all clear where it comes from -
> I'm starting to
> suspect that some context references belong in the zapatel.conf file.

Your suspicions are correct (although there is no "zapatel.conf" file).
Specifically, the file where you define context for Zaptel channels is
/etc/asterisk/zapata.conf. There are two "zap" files that are required
for Asterisk: /etc/zaptel.conf and /etc/asterisk/zapata.conf

/etc/zaptel.conf configures the Linux driver (the interface between the
hardware and Linux), whereas /etc/asterisk/zapata.conf defines the
Asterisk channel (the mechanism Asterisk creates to communicate with the
Zapata telephony interfaces).

Hierarchically, it goes something like this:

[Asterisk]
   |
[Asterisk Zapata channels (/etc/asterisk/zapata.conf)]
   |
[Linux Zapata driver (/etc/zaptel.conf)]
   |
[TDM Hardware]

Technically-speaking, the Zaptel cards do not need Asterisk to work.
They are devices under Linux, and could be used by any program designed
to work with them. That is why their configuration is stored in the /etc
directory.

But if Asterisk is to use the Zapata cards, it requires a configuration
file to define how it will interact with the hardware; it needs to have
those channels defined. That file, being specific to Asterisk, is
therefore located in the /etc/asterisk directory.

In reality, Asterisk is the only product making use of the Zapata
telephony interfaces, so the configuration can seem confusing. But it is
very appropriate when one understands the difference between the Zaptel
hardware driver, and Zapata channels in Asterisk.

> A comment about where the document leaves off.  In the
> beginning the document promises to get to a minimal working
> set, but it really doesn't go that far. Unless I've missed
> something, we aren't left with even a complete version of the
> minimal example extensions.conf file.  Something is missing

With apologies, this is true. That document is still very much at the
draft stage.

> so that I'm not
> getting a dial tone on the analog phone hooked up to the
> TDM11B and I have no
> idea why (can anyone clue me in?)  I also tried the:
> 
> 			[incoming]
> 			exten => s,1,Answer()
> 			exten => s,2,Playback(goodbye)
> 			exten => s,3,Hangup()
> 
> example and asterisk didn't appear to see the incoming call
> and answer the call
> at all.  I'd love for the example files to be complete enough
> that this example
> could actually work from either the external POTS line or
> even better an analog
> phone hooked to the FXS interface.

See below. I've hacked something together which should set you on the
right path.

> I think it would be great if attached to the document there
> was a "final"
> version of all of the config files which are known to work
> with the given
> configuration.

Excellent suggestion, and one which we will implement.

> Can you help get me to a dialtone on the internal side or an
> answer on the
> external side?

I'll do my best:

Try this in /etc/zaptel.conf:

fxoks=1
fxsks=4
loadzone=us
defaultzone=us

And for /etc/asterisk/zapata.conf you can try this:

; Zapata telephony interface
;
; Configuration file

[channels]

;let's set some parameters
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

caller_id="Zap1"
signalling = fxo_ks

;and assign them to a channel
channel => 1

; now we'll change some parameters
; (note that any parameters that have not been 
; changed will contunue to apply)
context=incoming
caller_id="Zap2"
signalling = fxs_ks

;and assign them to a channel
channel => 4


Finally, you'll want something along these lines in
/etc/asterisk/extensions.conf:

[general]

[default]
exten => 6123,1,Dial(Zap/1)
exten => 6444,1,Dial(IAX2/asterisk at switch-1.nufone.net/4569)
;
[incoming]

exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Background(demo-congrats)

include => default


This is pretty lean stuff, but it should help to get you going.


Regards,

Jim Van Meggelen
Asterisk Documentation Project


> Thanks
> Steve
> 
> 
> Leif Madsen wrote:
>> On Sat, 30 Oct 2004 12:18:20 -0400, Steve Totaro
>> <asterisk at totarotechnologies.com> wrote:
>> 
>>> Yes, it should be four unless you care to move the actual module on
>>> the card to the second slot.
>> 
>> 
>> I have fixed this in CVS now.  Should be propogated to the website
>> in a few minutes. 
>> 
>> While we do try and test everything, sometimes things get missed.
>> This is why getting people to test the configurations in Volume-One
>> and report back what does and does not work is important.
>> 
>> Thanks for pointing one out!
>> Leif Madsen.
>> http://www.asteriskdocs.org
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