[asterisk-users] Possible Bug? .call files executing multiple times

[Danny Nicholas]

Can you post the .call file (with called number blacked out) before call and
after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should
be from /v/s/a/o).

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brandon Phelps
Sent: Monday, August 29, 2011 8:45 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple
times

On 08/19/2011 09:14 AM, Brandon Phelps wrote:
> Hello all,
>
> We are setting up an auto-dialer to call customers based on the 
> opening of tickets in our internal ticketing system. Everything is 
> going fine so far except for one snag:
>
> To test the system we are implementing I am manually moving .call 
> files into the /var/spool/asterisk/outgoing directory like this:
>
> asterisk at dialerdev:~# cp test5703.call /tmp/test.call && mv 
> /tmp/test.call /var/spool/asterisk/outgoing/
>
> This works great and the call is immediately started, however more 
> often than not (ie. not all the time, but most of the time) after 
> answering the call or rejecting it (sending it to voicemail), another 
> call is performed using the same file.
>
> I notice that when a call is initiated the .call file is not removed 
> immediately. Instead, asterisk waits until the call is completed 
> before removing the call file, so it seems like 5-10 seconds into the 
> call since the .call file still exists another call is placed.
>
> Any advice on how we can avoid this situation and ensure that only one 
> call is made per .call file?
>
> The OS is Ubuntu 11.04 server and we're running Asterisk 1.8.
>
> Thanks,
>

Sorry to bring this back up but I am still having this issue and haven't had
any luck resolving it.  It should be noted that the .call files in question
are set to MaxRetries: 0, and simply connect the call to the 's' extension
in a custom context.  From there the context is pretty complicated, running
some AGI scripts along with some dealing with user input, basically a simple
IVR.

Any help would be appreciated.

Thanks,
Brandon

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users