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

[Richard Revels]

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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