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...OANSWER,n,GotoIf($["${IVR_RETVM}" != "RETURN" | "${IVR_CONTEXT}" = ""]?bye) exten => NOANSWER,n,Return exten => NOANSWER,n(bye),Macro(hangupcall) exten => TORTURE,1,Goto(app-blackhole,musiconhold,1) exten => TORTURE,n,Macro(hangupcall) exten => DONTCALL,1,Answer exten => DONTCALL,n,Wait(1) exten => DONTCALL,n,Zapateller() exten => DONTCALL,n,Playback(ss-noservice) exten => DONTCALL,n,Macro(hangupcall) ; make sure hungup calls go here so that proper cleanup occurs from call confirmed calls and the like ; exten => h,1,Macro(hangupca...

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

...answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [macro-stdPrivacyexten]; ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority) ; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)` ; exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databas...

...ntcode)}) exten=>_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg) exten=>_[123]xxx,n,NoOp(Dialstatus: ${DIALSTATUS}) exten=>_[123]xxx,n,GotoIf($["${DIALSTATUS}"="NOANSWER"]?takevoicemail:checkd ont) exten=>_[123]xxx,n(checkdont),GotoIf($["${DIALSTATUS}"="DONTCALL"]?takevoice mail:donecall) exten=>_[123]xxx,n,NoOp(Taking a voicemail...) exten=>_[123]xxx,n(takevoicemail),VoiceMail(${EXTEN}@default) exten=>_[123]xxx,n(donecall),Hangup() [call-redirect] include => parkedcalls exten=>_[123]xxx,1,NoOp(Transferring Call. This Channel ${CHAN...

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

...mailMain(${ARG1}) [macro-stdPrivacyexten] exten => s,1,Dial(${ARG2},20|p) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(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(...

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow

...o(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite "Don't call again" script. exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script. exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer e...