[asterisk-users] get SIP extension status without calling it
Richard Revels
rrevels at bandwidth.com
Sun Dec 2 18:40:55 CST 2007
I'm using 1.2.6 with the dialplan I posted so I guess the UA you are
using is just plain hosing you.
Anyway, with the queue I believe the music on hold is played to the
inbound side until the call is picked up by an agent. The queue tries
every <retry> seconds to get an agent for <timeout> seconds. If that
fails for however long you set the limit to then the dialplan
continues. I use this to set the priority of the call a little higher
and loop it back into the queue but it could be used for a problem
such as yours just as easily. I don't use the agent login and all
that so I may be talking about something that doesn't apply to your
configuration. If so, sorry for wasting your time.
exten => +1XXXXXXXXX,1,NoOp(Inbound call from ${CALLERIDNUM})
exten => +1XXXXXXXXX,n,Answer()
exten => +1XXXXXXXXX,n,Set(GROUP()=cloud)
exten => +1XXXXXXXXX,n,Set(QUEUE_PRIO=0)
exten => +1XXXXXXXXX,n(waiting),Queue(mainline,,,,600)
exten => +1XXXXXXXXX,n,Set(QUEUE_PRIO=$[${QUEUE_PRIO} + 5])
exten => +1XXXXXXXXX,n,GoTo(waiting)
exten => +1XXXXXXXXX,n,HangUp
On Dec 2, 2007, at 7:02 PM, Vieri wrote:
> I'd like to add that "show hints" on * CLI displays
> the following for ext 4053 tested below:
>
> 4053 : SIP/4053
> State:Idle Watchers 0
>
> (it should be "unavailable" or something, but anyway,
> ChanIsAvail reports an AVAILSTATUS of 0, ie. unknown)
>
> --- Vieri <rentorbuy at yahoo.com> wrote:
>
> > 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 although quite outdated).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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