[Asterisk-Users] TDM400P Newbie configuration :-)

[Joseph]

James Bean wrote:
> Sorry to post such a newb set of questions but I have been hammering
> about trying to get Asterisk running on FC2 machine reading everything
> available (I think that is what stuffed me, shouldn't have read it all
> :-) ).
> 
> Config
> 
> FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
> correctly.
> 
> Amazingly enough I have everything compiled correctly and installed.
> 
> I am running a TDM400P, Port 1 FXS, Port 4 FXO.
> 
> I have my PSTN line plugged into 1 port and my Analogue phone plugged
> into port 4 (I think that's right I get tone on the phone when I pick it
> up and echo works).
> 
> /etc/zaptel.conf
> 
> fxols=1
> fxsls=4
> ; Weird but I was told to have the fxols fxsls reverse to the actually
> module
> loadzone = au
> defaultzone = au
> 
> /etc/zapata.conf
> 
> [channels]
> context=default
> switchtype=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> signalling=fxo_ls
> callgroup=1
> pickupgroup=1
> immediate=no
> context=internal
> busydetect=yes
> callerid="James Bean<690>"      ;assuming extension 690
> mailbox=690                     ;stutter tone for voicemail - you can
> use an optional context here
> transfer=yes
> channel=>1
> group=2
> signalling=fxs_ls
> context=pstn

Here you have a context of pstn, which I assume is your incoming dialtone.

> channel=>4
> 
> Extensions.conf

But where is the pstn context in Extensions to match the above incoming 
dialtone?

Mayb you want something like this:

[pstn]

exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM})) ; Just put a 
comment in the CLI for info.
exten => s,2,Dial(Zap/g1,45,t)	;Dial the group=1 zap card mod above
exten => s,3,VoiceMail(u100)	;Whatever box you want.

> 
> [internal]
> exten => i,1,Playback(invalid)
> exten => i,2,Hangup
> exten => t,1,Hangup
> 
> exten => 099,1,Echo     ;simple echo test when you dial 099 on your
> phone
> 
> [outgoing]
> 
> exten => _1XX,1,Dial(H323/${EXTEN}@192.168.254.250)     ; 1xx extension
> to Salisbury
> exten => _2XX,1,Dial(H323/${EXTEN}@192.168.20.250)      ; 2xx extension
> to Marcoola
> exten => 610,1,Dial(H323/${EXTEN}@192.168.30.250)  ; 610 to Jindalee
> exten => 620,1,Dial(H323/${EXTEN}@192.168.40.250)  ; 620 to Batteryhill
> 
> exten => _54XXXXXX,1,Dial(H323/${EXTEN}@192.168.20.250) ; 54 to Marcoola
> exten => _0754XXXXXX,1,Dial(H323/${EXTEN}@192.168.20.250)    ; 54 to
> Marcoola
> 
> exten => _XXXXXXXX,1,Dial(Zap/g2/${EXTEN})
> 
> H323.conf
> 
> [general]
> port = 1720
> bindaddr = 192.168.69.1 
> tos=lowdelay
> 
> disallow=all
> allow=g723.1
> allow=gsm
> 
> --------------------------------------
> 
> I can pick up the phone and ring 099 and echo works but if I dial
> anything else I just get a busy signal with no errors on asterisk
> -vvvvc, what I need is for ANY incoming calls to make the analogue phone
> ring.

See comment above.

> 
> Outgoing calls that fit the rules use h323, everything else should pick
> up the PSTN line and dial.
> 
> I again apologise for the mess and newbness (did I just invent a word),
> I just need a kick start and get the basic stuff working before I start
> playing.
> 
> Also, anyone had asterisk talking to OKI Voip like BV1250 units
> working?, if so can you drop me an email.

No idea on that.

-- 

respectfully, Joseph
--------------------