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...oying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten => 10100,7,PlayBack(vm-goodbye) exten => 10100,8,Hangup ---<And Here>--- Normally when i call that extension if the user is online will ring if not, will play: "Extension is currently unavailable" and immediately should go...

...pe=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rules. I tried the "nat=yes" parameter but no changes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...

...n IAXCO1 ; exten => s,114,SetGlobalVar(FOUNDME=BUSY) exten => s,115,Goto(s-BUSY,1) ; ; returns here if busy on IAXCO2 ; exten => s,118,SetGlobalVar(FOUNDME=BUSY) exten => s,119,Goto(s-BUSY,1) ; exten => s-BUSY,1,BackGround(the-party-you-are-calling) exten => s-BUSY,2,BackGround(is-curntly-busy) exten => s-BUSY,3,SetGlobalVar(FOUNDME=BUSY) exten => s-BUSY,4,Goto(s-CLEANEXIT,1) ; exten => s-CANCEL,1,BackGround(canceled) exten => s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL) exten => s-CANCEL,3,Goto(s-CLEANEXIT,1) ; exten => s-CHANUNAVAIL,1,BackGround(channel) exten =>...

...intl] >> exten => s,1,ResetCDR(w) >> exten => s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000)) >> exten => s,3,Goto(s-${DIALSTATUS},1) >> exten => s,4,Hangup(19) >> exten => s-BUSY,1,NoCDR() >> exten => s-BUSY,n,Playback(user&is-curntly-busy) >> exten => s-BUSY,n,Hangup(19) >> exten => s-CONGESTION,1,NoCDR() >> exten => s-CONGESTION,n,Playback(user&is-curntly-busy) >> exten => s-CONGESTION,n,Hangup(19) >> exten => s-CHANUNAVAIL,1,NoCDR() >> exten => s-CHANUNAVAIL,n,Playba...

...}/w${ARG1},${ARG2}) exten => s,3,NoOp(${DIALSTATUS}) exten => s,4,Goto(s-${DIALSTATUS},1) ; ; Return here if busy ; exten => s,103,NoOp(${DIALSTATUS}) exten => s,104,Goto(s-${DIALSTATUS},1) ; ; exten => s-BUSY,1,BackGround(the-party-you-are-calling) exten => s-BUSY,2,BackGround(is-curntly-busy) exten => s-BUSY,3,SetGlobalVar(FOUNDME=BUSY) exten => s-BUSY,4,Goto(s-CLEANEXIT,1) ; ; exten => s-CANCEL,1,BackGround(canceled) exten => s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL) exten => s-BUSY,3,Goto(s-CLEANEXIT,1) ; ; exten => s-CHANUNAVAIL,1,BackGround(channel) exten =&g...

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

...R) exten => s,2,Dial(${PSTN}/w${ARG1},${ARG2}) exten => s,3,NoOp(${DIALSTATUS}) exten => s,4,Goto(s-${DIALSTATUS},1) exten => s,103,NoOp(${DIALSTATUS}) exten => s,104,Goto(s-${DIALSTATUS},1) exten => s-BUSY,1,BackGround(the-party-you-are-calling) exten => s-BUSY,2,BackGround(is-curntly-busy) exten => s-BUSY,3,SetGlobalVar(FOUNDME=BUSY) exten => s-BUSY,4,Goto(s-CLEANEXIT,1) exten => s-CANCEL,1,BackGround(canceled) exten => s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL) exten => s-CANCEL,3,Goto(s-CLEANEXIT,1) exten => s-CHANUNAVAIL,1,BackGround(channel) exten => s-C...

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

...r${HANGUPCAUSE},1) exten => _XXXXXX,4,Hangup() [dial-result] exten => r0,1,Hangup() exten => r1,1,playback(discon-or-out-of-service) exten => r1,2,Hangup exten => r16,1,Hangup() exten => r17,1,Busy() exten => r18,1,playback(the-number-u-dialed) exten => r18,2,playback(is-curntly-unavail) exten => r18,3,hangup exten => r27,1,playback(discon-or-out-of-service) exten => r27,2,hangup exten => r28,1,playback(that-is-not-rec-phn-num) exten => r28,2,hangup exten => r31,1,Congestion() exten => r34,1,Congestion() exten => r42,1,Congestion() exten =>...

...ter they're acknowledged I'd be glad to know. Zach #=================================================================== -- Goto (cynic-closed,s,1) -- Executing [s@cynic-closed:1] BackGround("IAX2/vitel-inbound-1", "normalized/technical-support&normalized/is-curntly-unavail") in new stack -- <IAX2/vitel-inbound-1> Playing 'normalized/technical-support' (language 'en') Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMF_E Subclass: 4 Timestamp: 05403ms SCall: 00023 DCall: 00001 [64.2.142.31:4569] Tx-Frame Retry[-0...