[asterisk-users] TDM410 PTSN line setup with 1 analog phone

[Lyle Giese]

I have not use a TDM4xx card for a while, but I remember that in order 
for ringing to work, you had to plug in an extra molex connector into 
the card to supply power to the ringing generator portion.

If you forgot to do that...

Lyle

BTW, I know about being a noobie.  I was there once myself and still am 
there every day learning and working with new stuff.  Sometimes not of 
my own choosing, but one must do what they need to keep getting those 
paychecks<GRIN>!

On 6/20/2012 8:44 AM, Joseph Towery wrote:
> Thanks Lyle,
>
> Sorry to sound so much like a newb but in asterisk I am.  I was
> initially trying to do things by hand in the extensions.conf file and
> had no luck.  I then got from SVN checkout asterisk-gui and used it to
> simply try and get things started, and created a trunk, users, incoming
> rule, etc. from the gui and finally got dial tone, and can dial out, but
> I haven't got the analog phone ringing yet.  I will have more targeted
> questions in the near future.  It is just hard to find "google" help for
> analog answers.  Most deal with SIP (which is my next step once I have
> the analog lines working).
>
> Thanks,
>
> ------------------------------------------------------------------------
> *From:* Lyle Giese <lyle at lcrcomputer.net>
> *To:* asterisk-users at lists.digium.com
> *Sent:* Tue, June 19, 2012 9:29:12 PM
> *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone
>
> An FXO port needs to be connected to dial tone or your PSTN line. And an
> FXS port needs to be connected to the station equipment(ie. a physical
> phone).
>
> The TDM410 is basically a channel bank to Asterisk, so the channel type
> inside Asterisk is FXO to talk to the physical FXS card and FXS to talk
> to the physical FXO port.
>
> Lyle Giese
> LCR Computer Services, Inc.
>
> On 06/18/12 15:08, Joseph Towery wrote:
>> 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 samples so I realize
>> that may have messed me up.
>>
>> This is all running on Ubuntu Server 12.04.  I have been
>> googling/researching reading the book, etc.  Everything I find is for
>> SIP softphones etc.  I just want to start by getting the asterisk
>> machine to provide dialtone to the analog phone, and ring that phone
>> when I call the PTSN line.
>>
>> I must be missing something in the basic dahdi and dialplan to simple
>> get the analog phone to work.  Can someone point me to a example of
>> what I am trying to accomplish?  Not wanting handholding but a push in
>> the right direction.
>>
>> Thanks.
>>
>>
>> --
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>
>
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