Solved

Test.CALL file in place but no action?

Posted on 2008-06-25
5
967 Views
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
0
Comment
Question by:tom_szabo
  • 2
  • 2
5 Comments
 
LVL 36

Expert Comment

by:grblades
Comment Utility
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.
0
 

Author Comment

by:tom_szabo
Comment Utility
Well I did, many times, no luck. Is there a module needs to be loaded? Anything I am missing?
0
 
LVL 36

Accepted Solution

by:
grblades earned 300 total points
Comment Utility
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.
0
 
LVL 25

Assisted Solution

by:kode99
kode99 earned 200 total points
Comment Utility
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.
0
 

Author Closing Comment

by:tom_szabo
Comment Utility
thanks
0

Featured Post

IT, Stop Being Called Into Every Meeting

Highfive is so simple that setting up every meeting room takes just minutes and every employee will be able to start or join a call from any room with ease. Never be called into a meeting just to get it started again. This is how video conferencing should work!

Join & Write a Comment

Almost all Internet protocol telephones have built-in switches at the back that allow you to connect your personal computer to one port and use the other port to connect your phone to to a Cisco switch.   Why we need to connect the PC to the pho…
As companies replace their old PBX phone systems with Unified IP Communications, many are finding out that legacy applications such as fax do not work well with VoIP. Fortunately, Cloud Faxing provides a cost-effective alternative that works over an…
Get a first impression of how PRTG looks and learn how it works.   This video is a short introduction to PRTG, as an initial overview or as a quick start for new PRTG users.
Here's a very brief overview of the methods PRTG Network Monitor (https://www.paessler.com/prtg) offers for monitoring bandwidth, to help you decide which methods you´d like to investigate in more detail.  The methods are covered in more detail in o…

762 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question

Need Help in Real-Time?

Connect with top rated Experts

7 Experts available now in Live!

Get 1:1 Help Now