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 <l...@lcrcomputer.net>
*To:* asterisk-users@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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