[asterisk-users] how to determine if a SIP extension has DNDonoroff

[Vieri]

--- "Eric \"ManxPower\" Wieling" <eric at fnords.org>
wrote:

> Anthony Francis wrote:
> > Eric "ManxPower" Wieling wrote:
> >> SIP response 486 is "Busy Here" according to RFC
> 3326.

According to http://www.faqs.org/rfcs/rfc3261.html the
SIP client can either send back a 480 or a 486 in case
DND is on:

21.4.18 480 Temporarily Unavailable
21.4.24 486 Busy Here
21.6.1 600 Busy Everywhere

In any case, if I reconsider my original post where
all I need to do is Add/Remove QueueMembers according
to the "state" of logged-on agents, how can I simply
retrieve the SIP response code? (from there on, my AGI
script will do whatever it deems fit)
For example, can "sip show peer X" be easily patched
so that I can get the response code?
Or would there be an easy way to set a "global"
variable with the latest response code of a particular
extension? (eg. could Asterisk keep track of all SIP
extension response codes?)

Otherwise, as another post suggested, could the Dial()
command be modified so that it "queries" the remote
peer just to see if it can establish a call or not and
thus return immediately with the state in DIALSTATUS?

I mean, there *must* be a way to get the client's SIP
response code from the * server but I don't know how.

Vieri



       
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