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

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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