[Asterisk-Users] Newbie X100P Clone question

[Seth Remington]

On Fri, 2004-11-05 at 15:58, Steve Frank wrote:
> 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


You've numbered both lines as priority 2. Fix that and you'll be fine.

-Seth


> 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
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-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559