search for: emergency_trunk

Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these extensions.conf lines: exten => s,1,SetVar(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,SetGlobalVar(EMERGENCY=1) exten => s,n,SetVar(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress) Now, I have several questions: * What is the "n" priority and how can...

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress) exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten => s,n,Wait(12) exten => s,n,Goto...

...(In use) (SWEC: MG2) 4 FXS FXOKS (In use) (SWEC: MG2) ### Span 2: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1" root at Trixie:~# ------------------------------------------------- extensions.conf (excerpt) ; Global variables [globals] ; Stuff for 911 EMERGENCY=0 EMERGENCY_TRUNK=DAHDI/4 ; Change this for production use: ;EMERGENCY_NUM=some_test_phone_number EMERGENCY_NUM=811 ;EMERGENCY_NUM=911 ... ; Which trunk to use for any DAHDI (PSTN-'Hard Line'-AKA POTS) type stuff POTSTRUNK=DAHDI/4 ... ; Emergency -- DO NOT REMOVE! exten => 911,1,Goto(nineoneone,s,1)...

...ONE4=SIP/24 ACAPHONE5=SIP/25 ACAPHONE6=SIP/26 ACAPHONE7=SIP/27 GMNETPHONE0=SIP/30 GMNETPHONE1=SIP/31 GMNETPHONE2=SIP/32 GMNETPHONE3=SIP/33 GMNETPHONE4=SIP/34 GMNETPHONE5=SIP/35 GMNETPHONE6=SIP/36 GMNETPHONE7=SIP/37 EXTERNPHONE0=SIP/150 CPHONE1=SIP/16780000000 CPHONE2=SIP/17700000000 EMERGENCY=0 EMERGENCY_TRUNK=DAHDI/G1 ; Change this for production use: EMERGENCY_NUM=6789542133 [from-pstn] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming1,s,1) [from-pstn1] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_...

...ng, te405p-in and sip need access to each other and the ability to dialout, and voip, voip needs access to dial the ericsson system and the sip phones (haven't added that part yet) but not access to an outside line. James My extensions.conf #include extensions_sip.conf [globals] EMERGENCY=0 EMERGENCY_TRUNK=Zap/10 [dialstring] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup [atp-out] exten => _9X.,1,Dial(IAX2/username:password@overhere.overthere/${EXTEN:1}) exten => _9X.,2,Congestion exten => _9X.,3,Hangup [atp-in] exten => 30182849,1,SetMusicOnHold(re...

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

...pass at iXcall register = user:pass at messagenet register = user:pass at 069929XXXX register = user:pass at 9083XXX extension.conf: [general] static = yes writeprotect = no autofallthrough = yes clearglobalvars = no priorityjumping = no [globals] TRUNKOPTIONS = EMERGENCY = 0 EMERGENCY_TRUNK = TRANSFERS_CTX = DefaultOutgoingRule CALLBACK_CTX = DefaultOutgoingRule DISA_CTX = DefaultOutgoingRule DISA_PASSWD = 1111 DYNAMIC_FEATURES = automon TRUNK = SIP/3124XXXXX&SIP/9083XXX&SIP/069929XXXX&SIP/webcalldirectDE&SIP/webcalldirectNL&SIP/iXcall&SIP/messagenet OUTGOING_...

...) exten => 388,4,Hangup exten => 390,1,playback(lm1/call_may_be_recorded) exten => 390,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/DNE|te405p-in,Zap/g4/211,1) [emergency] exten => s,1,Dial(ZAP/g1/000) exten => s,1,SetVar(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,SetGlobalVar(EMERGENCY=1) exten => s,n,SetVar(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress) exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten => s,n,Wait(12) exten =&...

...the ZAp channel: sipserver*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Extensions.conf: [general] static=yes writeprotect=no [globals] EMERGENCY=0 EMERGENCY_TRUNK=Zap/17 EMERGENCY_NUM=some_test_phone_number MAX_RING_TIME = 20 CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g1...

...505@fnords.org> Content-Type: text/plain; charset=us-ascii; format=flowed John Goerzen wrote: > Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these > extensions.conf lines: > > exten => s,1,SetVar(SET_EMERG_FLAG=0) > exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) > exten => s,n,SetGlobalVar(EMERGENCY=1) > exten => s,n,SetVar(SET_EMERG_FLAG=1) > exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) > exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress) I hope the wiki page mentions that the "n" priority...