[Asterisk-Users] setting SIP to dial PSTN with TDM400P

[Martijn van Oosterhout]

Hi,

I've never used fxs/fxo modules, only E1 cards so I'm not entirely
sure. However, this log:

> *CLI>     -- Starting simple switch on 'Zap/1-1'
>     -- Executing Dial("Zap/1-1", "Zap/1/6998256") in new stack
>     -- Called 1/6998256
>     -- Zap/1/6998256-busy-1013475805 is busy
>     -- Hungup 'Zap/1/6998256-busy-1013475805'
>   == Everyone is busy/congested at this time
>     -- Timeout on Zap/1-1
>   == CDR updated on Zap/1-1

seems to indicate you're making the call from Zap/1 and trying to make
the outgoing call on Zap/1 also. I think you need to figure out which
Zap channel is your FXO and which is your FXS. Maybe the outgoing is
Zap/2? "zap show channels" gives a list I beleive...

Secondly, your config files only seem to mention one channel. Have you
looked at Asterisk at Home. It seems to autodrtect your config somehow....

On Fri, Apr 01, 2005 at 01:08:24PM +0500, Muhammad Haris wrote:
> to dear martijn,
> 
> i made every possible change i can make.... 
> i have a TDM400P Zap card...
> i had connected PSTN line to FXO Kewlstart at channel 1.
> and analog phone to FXS Kewlstart at Channel 4.
> i can hear continous ring tone when i hook up the receiver.
> plz have a look at my confs.

Have a nice day,
-- 
Martijn van Oosterhout
Ecomtel Pty Ltd