[Asterisk-Users] Outbound calling with TDM400P

[Simon Brown]

Try removing the g from the dial command:

exten => _X.,1,Dial(Zap/1/${EXTEN},60) 
exten => _X.,2,Hangup ;
exten => _NXXXXXXX,1,Dial(Zap/1)

Simon Brown

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Tarte
Sent: Wednesday, 2 February 2005 16:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outbound calling with TDM400P

A little more investigation:

I hooked up another phone to a splitter so I could listen to the outbound
line.  There are no sounds of any sort coming out on the line when the FXO
should be dialing.  I put some debug in the zaptel driver, and I can see the
driver trying to dial.  It calls __do_dtmf() with all of the digits that I
would like it to dial, but there is no sound on the wire.  Any ideas?

Thanks,

Rob

Rob Tarte wrote:

> I am trying to place an analog outbound call from a Sipura SPA-841 
> through a * server with a TDM400P and 4 FXO's.  When I call in from an 
> analog line everything works fine, I can talk over the SIP phone.
> When I call out, * says:
>
> == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 
> 'SIP/sipphone-d29d'
> -- Executing Dial("SIP/sipphone-9eb0", "Zap/g1/[phonenumber]|60") in 
> new stack
>  -- Called g1/[phonenumber]
> -- Zap/1-1 answered SIP/sipphone-9eb0
>
> And then I get silence.  The phone doesn't ring on the other end.  I 
> have attached my configuration files.
>
> Any help would be greatly appreciated,
>
> Rob
>
> ------------------------------------- sip.conf
> ----------------------------
> [general]
> context=default
> port=5060
> bindaddr=0.0.0.0
> srvlookup=yes
>

>
> [sipphone]
> type=friend
> context=from-sip
> username=sipphone
> fromuser=sipphone
> callerid=Incoming Call<101>
> host=dynamic
> nat=no
> canreinvite=yes
> dtmfmode=info
> incominglimit=1
>

>
> mailbox=101 at default
> disallow=all
> allow=ulaw
>

>
> allow=alaw
> allow=g723.1
> allow=g729
>

>
> -------------------------------- zaptel.conf ----------------------- 
> loadzone = us defaultzone=us
> fxsks=1-4
>

>
> -------------------------------- zapata.conf -----------------------
>

>
> [channels]
> switchtype=national
> rxwink=300              ; Atlas seems to use long (250ms) winks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> callerid=asreceived
>

>
> group=1
> signalling=fxs_ks
> languange=en
> context=default
> channel => 1-4
>

>
> -------------------------------- extensions.conf 
> ----------------------- [general] static=yes writeprotect=no
>

>
> [globals]
> IAXINFO=guest                                   ; IAXtel 
> username/password
> OUTGOING => Zap/1
>

>
> [from-sip]
> ignorepat => 9
> exten => _X.,1,Dial(Zap/g1/${EXTEN},60) exten => _X.,2,Hangup ;exten 
> => _NXXXXXXX,1,Dial(Zap/g1)
>

>
> [default]
> exten => s,1,Wait,1                     ; Wait a second, just for fun
> exten => s,2,Answer                     ; Answer the line
> exten => s,3,Dial(SIP/sipphone)
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--
Robert Tarte
Pacific CodeWorks
1347 Pacific Ave., Suite 202
Santa Cruz, CA 95060

(p) 831-426-7582
(f) 831-426-7584

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