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

[Vieri]

Thanks Richard but I think that ChanIsAvail must be
buggy (based on some user comments in the wiki,
although quite outdated).

I have the hint entry as you say (am using FreePBX and
it's already there).

But whenever I call ChanIsAvail with the s option I
always get:
${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN - "Unknown";
channel is valid, but unknown state. 

I might be doing something wrong but here is the code:

[IVR-menu1]
exten => s,1,Answer()
(...)
exten => s,n,Playback(welcome)
exten => s,n,ChanIsAvail(SIP/4053|s)
exten => s,n,NoOp(DEBUG: availstatus is
${AVAILSTATUS})

In extensions.conf I also have:
exten => 4053,hint,SIP/4053

I'm using Astrisk 1.2. Is ChanIsAvail working well in
1.2?

As far as setting a time limit on a call in the queue
is concerned, it doesn't sound "nice" for the caller
to be dropped after a few rings when it could have
been dropped right fom the beginning. It could be a
solution but it doesn't sound "right" ;-).

Vieri

--- Richard Revels <rrevels at bandwidth.com> wrote:

> In the sip.conf entry assign a context.
> 
> In that context, hint the extension i.e. exten =>
> 7302,hint,SIP/7302.
> 
> Before you get ready to dial, or whatever, do
> chanisavail  i.e.
> 
> exten =>
> _1XXXX,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
> exten => _1XXXX,n,Playback(beep)
> exten => _1XXXX,n,Dial(SIP/${EXTEN},2)
> exten =>
> _1XXXX,n,Goto(result-${DIALSTATUS},${EXTEN},1)
> exten => _1XXXX,CheckUse+101,SayDigits(${EXTEN:1})
> exten => _1XXXX,CheckUse+102,Playback(vm-isonphone)
> exten => _1XXXX,CheckUse+103,Hangup()
> 
> This is from the paging stuff.  It checks the
> primary extension before  
> ringing the auto answer extension of the phone.  I
> seem to remember it  
> detecting DND as well for the Cisco 7960.
> 
> I don't see it in this message but I seem to
> remember seeing somewhere  
> in this thread that the goal is to keep people from
> being in a queue  
> forever.  Why not just set a time limit on the queue
> and play back  
> "all operators busy" and hang up if a call hits that
> limit?
> 
> Richard
> 
> 
> 
> On Dec 2, 2007, at 8:51 AM, 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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