asterisk users - Jun 2004 - Getting Asterisk to automatically dialout

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.

On Tue, 29 Jun 2004, Andrew Elchuk wrote:
> I'm trying to get asterisk to auto-dail out.  I created a *.call file
did you create the file in /var/spool/asterisk/outgoing/, or did you
create it elsewhere and then move it to that directory? The docs mention
that if the file is created in the outgoing directory, * may read the file
before you've finished writing it. But if you create the file elsewhere
and then move it, the entire contents are guaranteed to be in the
filesystem when * finds it. This might explain the difficulty you're
having..

Greg

On Tue, 2004-06-29 at 12:10, 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:
First, don't call out on specific channels. Put the channels in a group
and then use the group. Even if right now you only have one channel, it
will help you out later.

The first type of call file would be like this below
##############
Channel: Zap/g1/2609944
MaxRetries: 2
RetryTime: 60
WaitTime: 30

# now to a context/extension/priority that does what you want.
Context: AutoAnswer
Extension: 4
Priority: 1
###################
> [outgoing]
> 
> exten => s,1,Wait(1)
> exten => s,2,Dial(Zap/1/2609944)
This would never work since the dial would connect out and not return
till the call hungup.


-- 
Steven Critchfield <critch@basesys.com>

You don't need the Dial command in extensions.conf for outbound calls.
Asterisk will do the dialout based on your .call file

regards
Scott Stingel
 


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com 

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Andrew Elchuk
Sent: Tuesday, June 29, 2004 10:11 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Getting Asterisk to automatically dialout

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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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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> Asterisk-Users@lists.digium.com
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