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Test.CALL file in place but no action?

Posted on 2008-06-25
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Last Modified: 2013-12-21
As suggested by others I have installed Samba on the Elastix box and set up the share. So now I can drom a CALL file into the /var/spool/asterisk/outgoing folder. This works, but the file doesn't do anything! I mean Asterisk is not doing anything:
Channel: SIP/211
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: from-internal
Extension: 6173847XXXX
Priority: 1
AlwaysDelete: Yes

It doesn't even get deleted. Time is not in the future.

Any clues?

TIA

Tom
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Question by:tom_szabo
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Expert Comment

by:grblades
ID: 21872664
Check to make sure that you are creating the file elsewhere and then move or copy it into the outgoing directory. Otherwise if the file is still being written when asterisk tries to read it then it will fail.
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by:tom_szabo
ID: 21873032
Well I did, many times, no luck. Is there a module needs to be loaded? Anything I am missing?
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grblades earned 300 total points
ID: 21873090
See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

I would try creating the file in a different directory on the server.
Then edit it you make sure the line formatting is correct. Windows uses CRLF as line terminators while linux just uses LF. It shouldnt cause a problem but you never know.
Once you have the file move it to the outgoing directory (dont copy) and then see if it works.
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Assisted Solution

by:kode99
kode99 earned 200 total points
ID: 21879498
Given that you are dealing with an appliace type Asterisk I would not be surprised if it does not automatically load everything - only what it needs for the intended functionality.  

Check your modules.conf file,  two things to check.  The module needed is the pbx_spool.so.

If it has autoload=yes then also check that the pbx_spool module is not excluded with 'noload =>'.

Or if autoload=no then you may need to add a load=pbx_spool.so or uncomment the one that may be there.  

Of course make sure the is a pbx_spool.so module to load.

Here's some modul.conf examples,

http://www.voip-info.org/wiki/view/Asterisk+Slimming

I am not sure about how exact the formatting needs to be but I agree with grblades,  you want to check it closely.
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Author Closing Comment

by:tom_szabo
ID: 31470860
thanks
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