[Asterisk-Users] Getting Asterisk to automatically dialout

[Andrew Elchuk]

I tried putting a "callprogess=yes" above the "channel => 1" line in 
zapata.conf.  I then created a call with a *.call file it got 'stuck' at 
the "Dialing" state.  It then reported an error saying the call couldn't 
go through.  This is output I got from the asterisk CLI:

-- General --
Name: Zap/1-1
Type: Zap
UniqueID: 1088544582.2
Caller ID: Nagios
DNID Digits: (N/A)
State: Dialing (3)
Rings: 0
NativeFormat: 68
WriteFormat: 4
ReadFormat: 4
1st File Descriptor: 16
Frames in: 278
Frames out: 0
Time to Hangup: 0
--   PBX   --
Context: incoming
Extension: s
Priority: 1
Call Group: 0
Pickup Group: 0
Application: (N/A)
Data: (None)
Stack: -1
Blocking in: ast_waitfor_nandfds
*CLI> Jun 29 15:30:14 NOTICE[139279]: pbx_spool.c:232 attempt_thread: 
Call failed to go through, reason 0

In the call file I created after it connects to Zap/g1/2609944 it should 
go to the alert context of extensions.conf.  But after I put in the 
"callprogess=yes" line it seems to be getting hungup at the dialing 
state and it is using the incoming context??  Could this be a reason why 
it won't dial out for me?

Andrew Elchuk wrote:

> Hi,
> I'm trying to get asterisk to auto-dail out.  I created a *.call file 
> with the the top of it being "Channel: Zap/1/2609944", which should 
> have connected to Zap channel 1 and dial out to 2609944, but It did 
> not do so, asterisk would say a call was completed to Zap/1/2609944 
> but I never heard that phone ring.  So I tried just putting "Channel: 
> Zap/1" at the top of the call file so it would connect to Zap channel 
> 1, then in the *.call file connect it to an "outgoing" context in 
> extensions.conf which looked like:
> [outgoing]
>
> exten => s,1,Wait(1)
> exten => s,2,Dial(Zap/1/2609944)
> exten => s,3,Wait(2)
> exten => s,4,Playback(soundfile)
> exten => s,5,Hangup
>
> But when it ran this, asterisk told me it was unable to create a 
> channel of type "Zap", but then that a call was still completed to 
> Zap/1.  I've read everything about auto-dialout on voip-info.org and 
> read digium faqs and everything and have been unable to find a 
> solution.  If someone out there has had a similar problem and figured 
> it out or knows what might be wrong with what I'm trying to do it 
> would be greatly appreciated if you could help me out.  Thanks.
>
>
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