<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>Thanks Lyle,<br><br>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).<br><br>Thanks,<br></div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><br><div style="font-family:times new roman, new york, times,
serif;font-size:12pt"><font face="Tahoma" size="2"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Lyle Giese <lyle@lcrcomputer.net><br><b><span style="font-weight: bold;">To:</span></b> asterisk-users@lists.digium.com<br><b><span style="font-weight: bold;">Sent:</span></b> Tue, June 19, 2012 9:29:12 PM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone<br></font><br>
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).<br>
<br>
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.<br>
<br>
Lyle Giese<br>
LCR Computer Services, Inc.<br>
<br>
On 06/18/12 15:08, Joseph Towery wrote:
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<div>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. <br>
<br>
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.<br>
<br>
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.<br>
<br>
Thanks.<br>
</div>
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