Thanks for the reply!   Unfortunately, getting rid of the g didn't change anything.  I'm wondering if the tones aren't getting downloaded correctly to the driver or something.  The driver seems to be going through the motions, but there is no sound.  Any other ideas?

Thanks again,

Rob

Simon Brown wrote:
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] 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

    

  
[EMAIL PROTECTED]
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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