search for: _9xxxxxxxxxxx

...=> 888800,2,Voicemail(1000) ; exten => 769611,1,Macro(oneline1,${FSEXT1}) exten => 769615,1,Macro(oneline1,${LONDONSOLE1}) ;exten => 769616,1,Macro(oneline1,${LONDONSOLE2}) exten => 769636,1,Macro(oneline1,${FSEXT1},${401}) ;exten => 769637,1,Macro(oneline1,${NIGEL}) ; exten => _9xxxxxxxxxxx,1,Dial,(${OUTBOUND}/${EXTEN:1}) exten => _9xxxxxxxxxxx.,2,Congestion() exten => _9xxxxxxxxxxx,102,Congestion() ; exten => 999,1,Dial,(${OUTBOUND}/999) exten => 9999,1,Dial,(${OUTBOUND}/999) ; [root@asterisk asterisk]# more zapata.conf [channels] language=en usecallerid=yes hidecallerid=...

.... How do I tell * to stop listning for the DTMF? Here is my configs: CVS head 7/14/2004 The call is made Zap to Zap no sip involved. T100P t-1 (goes to provider ls t1) T100P t-1 (goes to NEC pbx) exten listing for outbound ld: TRUNK=Zap/g2 ; dial a long distance outbound number exten => _9XXXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},,Tt) exten => _9XXXXXXXXXXX,2,Congestion ; Timeout and invalid rules exten => #,1,Playback(invalid) exten => #,2,Hangup exten => t,1,Goto(#,1) exten => i,1,Playback(invalid) Any ideas? t o n y

...problem, but when users on Asterisk server 2, dial 9XXX-XXX-XXXX the call never reaches the zap channel on Asterisk server 1. I have a workaround working right now using switch => but I think there should be a better way to do this. Asterisk Server 2 -- extensions.conf [outbound] exten => _9XXXXXXXXXXX,1, Dial(SIP/outbound-server/${EXTEN:1},30,r) exten => _9XXXXXXXXXXX,2, HangUp() Asterisk Server 2 -- sip.conf [outbound-server] type=friend username=outbound-server secret=******** context = all-calls host=1.1.1.1 nat=no canrei...

...6009|1) exten = s,1,Goto(default|6009|1) [numberplan-custom-1] plancomment = DialPlan1 include = default exten = _9XXXXXXX,1,Macro(trunkdial,${TRUNK}/${EXTEN:1}) comment = _9XXXXXXX,1,Local,standard exten = ,1,Macro(trunkdial,${TRUNK}/${EXTEN:0}) comment = ,1,Emergency911,custom exten = _9XXXXXXXXXXX!,1,Macro(trunkdial,${TRUNK}/${EXTEN:1}) comment = _9XXXXXXXXXXX!,1,Longdistance,standard Date: Thu, 31 Jan 2008 09:04:03 -0500 From: "C F" <shmaltz at gmail.com> Subject: Re: [asterisk-users] Default delay time for Attended call transfer To: "Asterisk Users Ma...

Let say I have a server located in Europe and one in North America. The 2 servers are connected together with iax2. Both server are connected to phone lines in there own country. If I want that when a user call a north american phone number from the server in Europe it use a zap channel on the server located in North America and also if someone in North America dial an European phone