Hey, I would also recommend to use SIPPEER and with that verify the status of said peer. Based on that status, make the dialling decision. If you want more help, contact me directly.
Rennes Neps Elion Ettevõtted AS tel: +372 6402183 mob: +372 56490388 [email protected] -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Phil Frost Sent: Wednesday, March 21, 2012 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] fallback to default extension On Mar 21, 2012, at 08:36 , Andrew Latham wrote: > On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino <[email protected]> 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
