[Asterisk-Users] Newbie X100P Clone question

[Steve Frank]

I've got an X100P, I'm brand new to Asterisk. I've been able to set up
SIP extensions and have them working, now I've added the X100P in so I
can drop a line in and eventually be my outside world connection.
 
I've downloaded the zaptel code via CVS, and configured it up pretty
much exactly like
http://www.digium.com/index.php?menu=configuration#X100P by adding items
into my configurations.  My current extensions.conf looks like this:
 
[general]
static=yes
writeprotect=yes
 
[bogon-calls]
exten => _.,1,Congestion
 
[default]
exten => _XXXXXX,1,Dial,Zap/1/${EXTEN} ; Press any 7 digit number and
try to dial that number through Zap channel 1
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,2,Playback(demo-congrats) ; Plays the demo-congrats file
after answering the line
exten => s,3,Hangup
 
[from-sip]
exten => 3073,1,Dial(SIP/3073,20)
exten => 3073,2,Voicemail(u3073)
exten => 3073,102,Voicemail(b3073)
exten => 3073,103,Hangup
 

exten => 3087,1,Dial(SIP/3087,20)
exten => 3087,2,Voicemail(u3073)
exten => 3087,102,Voicemail(b3073)
exten => 3087,103,Hangup
 
exten => 3089,1,Dial(SIP/3089,20)
exten => 3089,2,Voicemail(u3089)
exten => 3089,102,Voicemail(b3089)
exten => 3089,103,Hangup
 

exten => 3123,1,VoicemailMain(${CALLERIDNUM})
 
Here's zapata.conf:
 
[trunkgroups]
 
[channels]
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
busydetect=no
callprogress=no
callerid=asreceived
group=1
context=default
channel => 1
 
An inbound call to the extension doesn't play back the "congrats" demo
gsm recording.  Running asterisk with -vvvvgc I get the following upon
dial in:
 
 
*CLI>     -- Starting simple switch on 'Zap/1-1'
    -- Executing Wait("Zap/1-1", "1") in new stack
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
    -- Starting simple switch on 'Zap/1-1'
Nov  5 14:54:59 WARNING[1967]: chan_zap.c:5466 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
    -- Executing Wait("Zap/1-1", "1") in new stack
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'

What's up with the exited non-zero on the spawn extension?  
 
Also, whenever starting Asterisk I always get this about 10 seconds
after init:
 
Nov  5 14:54:45 NOTICE[1958]: pbx_dundi.c:2841 destroy_trans: Peer
'00:50:8b:f3:75:bb' has become UNREACHABLE!
 
What does that mean?

 
Thanks very much in advance. This setup is very very interesting when
compared to our current production Interactive Intelligence CIC
system....

Steve