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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