[asterisk-users] get SIP extension status without calling it

[Vieri]

Thanks for the "sip show peers" script, Dave.
But that won't work for me.
It won't tell me whether the extension will actually
accept a call or not (eg. if DND is ON only on the
"client side").

This link might clarify the problem I am facing:

http://lists.digium.com/pipermail/asterisk-users/2007-September/195936.html

and the following links discuss a way to determine an
extension's DND state in order to use the
{Add,Remove}QueueMember function efficiently from a
custom cron script.

http://lists.digium.com/pipermail/asterisk-users/2007-September/196345.html

http://lists.digium.com/pipermail/asterisk-users/2007-September/196437.html

The need to determine if an extension accepts calls or
not (and what's missing here is to detect DND on/off
on the client side) is related to queues and agents.
Basically, if, say, all agents are in the queue but
have DND on then what I need is to bail the caller out
because it doesn't make much sense from a practical
point of view to have he/she wait "forever" for an
agent to turn DND off.

Maybe it's a big limitation in SIP protocol but I'd
like to know if other users have found a viable, open
source solution.

--- dave cantera <david.cantera at iacnet.net> wrote:

> vieri,
> you can get sip status with the following shell
> script...   I named it 
> 'sipshowpeer'...

> Vieri wrote:
> > Hi,
> >
> > I am trying to get a SIP extension's status
> without
> > actually making a call.
> >
> > I am using sofia-sip's "options" example utility
> and
> > the sip clients are SJphone softphones.
> >
> > >From Asterisk I run the "options" utility and
> query a
> > sip extension at 10.215.147.240. I get:
> >
> > # ./options -1 --all sip:10.215.147.240
> > SIP/2.0 501 Not Implemented
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27
> > From: <sip:10.215.144.27>;tag=U3DKgF7HgFKXH
> > To: "unknown" <sip:10.215.147.240>;tag=614733430
> > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472
> > CSeq: 92182805 OPTIONS
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > I guess that the softphone should be answering
> with a
> > 2xx code followed by a status description?
> > So I tried with the INVITE method and set DND on
> the
> > SIP extension:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 486 Busy Here
> > Via: SIP/2.0/UDP
> >
>
10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27
> > From: <sip:10.215.144.27>;tag=590Z1ND8B6XpN
> > To: "unknown" <sip:10.215.147.240>;tag=1a2d77b524
> > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472
> > CSeq: 92182952 INVITE
> > Content-Length: 0
> > Server: SJphone/1.65.377a (SJ Labs)
> >
> > The above would suit me fine because I get a "486
> Busy
> > Here" response.
> > However, if DND is off then I get:
> >
> > # ./options -1 --all --method INVITE
> > sip:10.215.147.240
> > SIP/2.0 180 Ringing
> >
> > and the SIP extension actually "rings", as
> > expected.(but this is undesireable)
> >
> > Now, does someone know another way to get the
> status
> > (ie. does it accept calls or not?) without making
> the
> > extension "ring"?
> >
> > Thanks
> >
> > Vieri



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