Thanks Richard but I think that ChanIsAvail must be buggy (based on some user comments in the wiki, although quite outdated).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 ${AVAILSTATUS}) In extensions.conf I also have: exten => 4053,hint,SIP/4053 I'm using Astrisk 1.2. Is ChanIsAvail working well in 1.2? As far as setting a time limit on a call in the queue is concerned, it doesn't sound "nice" for the caller to be dropped after a few rings when it could have been dropped right fom the beginning. It could be a solution but it doesn't sound "right" ;-). Vieri --- Richard Revels <[EMAIL PROTECTED]> wrote: > In the sip.conf entry assign a context. > > In that context, hint the extension i.e. exten => > 7302,hint,SIP/7302. > > Before you get ready to dial, or whatever, do > chanisavail i.e. > > exten => > _1XXXX,n(CheckUse),ChanIsAvail(SIP/${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 ____________________________________________________________________________________ Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users