[asterisk-users] fallback to default extension

Bryant Zimmerman BryantZ at zktech.com
Wed Mar 21 15:56:45 CDT 2012


Minor Correction
Hi

I've pretty much have it setup properly with the following:
exten => _24XX,1,Dial(SIP/${EXTEN},30)
exten => _24XX,n,GotoIf($${DIALSTATUS}"="CHANUNAVAIL?noconn:conn)
exten => _24XX,n(noconn),GotoIf($["${EXTEN}"="2400"]?conn:force)
exten => _24XX,n(force),Dial(SIP/2400)
exten => _24XX,n(conn),hangup()

The only problem is that if 2400 rejects the call asterisk tries to
call extension 2400 again...
What am I doing wrong and how do I fix it?

TIA
Paolo

On Wed, Mar 21, 2012 at 1:44 PM, Phil Frost <phil at macprofessionals.com> 
wrote:
> On Mar 21, 2012, at 08:36 , Andrew Latham wrote:
>> On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino <paolo.supino at gmail.com> 
wrote:
>>> Hi
>>>
>>>  I was asked by our development departement to setup asterisk in a
>>> manner that if someone calls an extension in the department that was
>>> was only configured, but a handset was never attached to it to fall
>>> back to a default extension. For example: Someone calls extension
>>> 2408, but there's no phone attached to 2408 it should fall back and
>>> ring at 2400..
>>>
>>> How do I setup asterisk to find out if there's a phone attached to an
>>> internal number if not ring another extension?
>>>
>>
>> Just add a dial(SIP/2400) at a later priority or any of the other many
>> ways.  Assuming 2400 is you operator then set the var and drop to the
>> operator. Verify your options to you dial syntax and any std-exten
>> setups.
>
>
> You might want to additionally inspect ${DIALSTATUS} to know more about 
why the first Dial() (to 2408, in your example) failed, and then use the 
ExecIf or GotoIf applications to take different actions.
>
> You might also try the function SIPPEER, again coupled with ExecIF or 
GotoIf.
>
> --
> _____________________________________________________________________
> -- 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

--
_____________________________________________________________________
-- 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/20120321/f5588f0b/attachment.htm>


More information about the asterisk-users mailing list