[asterisk-users] Callback through extensions.conf?

[Sherwood McGowan]

Real quick, please respond to my question about where the callfile ends up
after a few minutes, as well as the modification time and the permissions on
the file ;-) These are good bits to know

On Mon, Feb 7, 2011 at 10:46 AM, Gilles <codecomplete at free.fr> wrote:

> On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards
> <asterisk.org at sedwards.com> wrote:
> >>      sudo /usr/sbin/asterisk -d -d -d -n -v -v -v
> >
> >Oops. A '-c' should be in there :)
>
> Thanks Steve for the help.
>
> I launched * with "asterisk -d -d -d -n -v -v -v -c", and ran "module
> show" to check that pbx_spool.so is loaded:
> =============
> *CLI> module show like pbx_spool.so
> Module                         Description Use Count
> pbx_spool.so                   Outgoing Spool Support 0
> 1 modules loaded
> =============
>
> Next, I moved the following callfile to /var/spool/asterisk/outgoing:
> =============
> #callfileSIP.call
> Channel: SIP/xlite
> Context: callback-dialtone-auth
> Extension: s
> Priority: 1
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> =============
>
> Asterisk runs as root, and owns this file as well.
>
> Unfortunately, nothing shows up in the console, the xlite extension
> isn't called, even after waiting for a few minutes.
>
> Could it be that pbx_spool.so isn't really loaded, or is Asterisk
> somehow configured to ignore callfiles?
>
> Thank you.
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110207/a446040c/attachment.htm>