asterisk users - Jun 2012 - TDM410 PTSN line setup with 1 analog phone

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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120618/3241d764/attachment.htm>

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.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120619/20f89aff/attachment.htm>