search for: availstatus

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 T...

...EXT ${EXTEN}:); // NoOp(DIALSTATUS=${DIALSTATUS}); // Necessary for Phoenix NoOp(DIALEDTIME=${DIALEDTIME}); // NoOp(HANGUPCAUSE=${HANGUPCAUSE}); // NoOp(AVAILSTATUS=${AVAILSTATUS}); // if (${DIALSTATUS} = NOANSWER && ${DIALEDTIME} >= 7) // make sure it actually rang { &AutoLogout(${EXTEN},${DIALSTATUS}); }...

Hello list. ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ?? ... exten => 1,n,ChanIsAvail(SIP/sin10) exten => 1,n,NoOp(chanisavail == ${AVAILSTATUS}) ... [Mar 4 15:10:16] -- Executing [1 at sin:7] ChanIsAvail("IAX2/testlocal-14088", "SIP/sin10") in new stack [Mar 4 15:10:16] -- Execu...

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the first part of the test with quotes, too, like this: exten => s,n,GoToIf(["${AVAILSTATUS}" = "1"]?autoanswer:fail) Leaving aside the completely separate madness of trying to determin...

...ere is a way to using page/intercom with normal ring and not with auto-answer? My dialplan: [ext-paging] include => ext-paging-custom exten => PAGE4441,1,GotoIf($[ ${CALLERID(number)} = 4441 ]?skipself) exten => PAGE4441,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL) exten => PAGE4441,n,Set(AVAILSTATUS=not checked) exten => PAGE4441,n,Goto(SKIPCHECK) exten => PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js) exten => PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state = ${AVAILSTATUS}) exten => PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6) exten => PAGE444...

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten => _2XX,4,SIPAddHeader(Call-Info: <sip:83.222.226.126>\;answer-after=0) exten => _2XX,5,Page(SIP/winsor_${EXTEN}) I found a mention of the problem here however, the patch that was recommended gave the same...

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten => s,n,NoOp(${AVAILORIGCHAN}) exten => s,n,NoOp(${...

...16s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: XXXXXXXXXXX Extension: XXXXX Priority: XXXXXX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=XXXXXXXXX SIPURI=sip:303 at XXXXXXXXXXXXXXXXX CDR Variables: level 1: clid="Ext. 303" <303> level 1: src=303 level 1:...

...u get it to roll over to ex2? This does NOT work because it rings all extensions: Dial(SIP/Phone1ex1&SIP/Phone2ex1&SIP/Phone3ex1&SIP/Phone1ex2&SIP/Phone2ex2&SIP/Phone3ex2) ChanIsAvail also does NOT work because it returns that the channel is available even if it is busy and ${AVAILSTATUS} is always 0. A natural idea would be ChanIsBusy(SIP/Phone1ex1&SIP/Phone1ex2) but no such command exists. Anyone know how to accomplish what I'm after? -- John Lange

...- Other line (not paging line...we don't want to disturb their other line) ; exten => s,1,ChanIsAvail(${ARG2}|js) ; j is for dump and s is for ANY call. Check to see if line 1 is available ;exten => s,n,Set(_ALERT_INFO="RA") ; This is for the PolyComs exten => s,1,NoOp(${AVAILSTATUS}) exten => s,n,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the Snoms and Others exten => s,n,NoOp() ; Add others here exten => s,n,Dial(${ARG1}||) exten => s,n,Hangup exten => s,102,Hangup() [page] ; Paging context ; Two line phone set to auto answer exten => 2201_com...

Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

...o see if SIP device is in use and DO NOT PAGE if they are ; ${ARG1} - Device to page ; ${ARG2} - Other line (not paging line...we don't want to disturb their other line) ; exten => s,1,ChanIsAvail(${ARG2}|js) ; j is for dump and s is for ANY call to indicate busy exten => s,n,NoOp(${AVAILSTATUS}) exten => s,n,SIPAddHeader("Alert-Info: <Bellcore-dr5>") ; This is the shortest ringer for the Cisco phones exten => s,n,SIPAddHeader(Call-Info:\;answer-after=0) ; For Linksys SPA exten => s,n,NoOp() ; Add others here exten => s,n,Dial(${ARG1}||A()) exten => s,n,Ha...

.../10-0820d8e0", "") in new stack == Spawn extension (default, 15, 3) exited non-zero on 'SIP/10-0820d8e0' astbox*CLI> What I am looking for is to play Playback(pbx-invalid) if a user enters a sip extension not created. I've been testing a few options using DIALSTATUS, AVAILSTATUS and their values but without luck as if the sip phone 11 is not registered the pbx-invalid message. Thansk for reading and any suggestion will be welcome. Richard -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! -------------- next part -------------- An HT...

...oicemail(b${ARG1}) exten => s-BUSY,2,Goto(default,s,1) exten => s-DONTCALL,1,Goto(${ARG3},s,1) exten => s-TORTURE,1,Goto(${ARG4},s,1) exten => _s-.,1,Goto(s-NOANSWER,1) exten => a,1,VoicemailMain(${ARG1}) [macro-page] exten => s,1,ChanIsAvail(${ARG1}|js) exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) exten => s,n(autoanswer),Set(_ALERT_INFO="RA") exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) exten => s,n,NoOp() exten => s,n,Dial(${ARG1}||) exten => s,n(fail),Hangup [page] exten => _X.,1,Macro(page,SIP/${EXTEN}) [default] e...

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

...ipt. exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [macro-page]; exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for Jump and s is for ANY call exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others exten => s,n,NoOp() ; Add others here and Post on the Wiki!!...