[Asterisk-Users] Outbound calling with TDM400P

Rob Tarte rtarte at pacificcodeworks.com
Tue Feb 1 10:35:45 MST 2005


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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