search for: phone1vm

...OIP provider until I can get the local asterisk setup to work. My config is below Thanks, Paul sip.conf -------- : [201] type=friend host=dynamic username=201 secret=<PW> mailbox=201 context=local callerid="Me" <201> extensions.conf --------------- [globals] PHONE1=SIP/201 PHONE1VM=201 : [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; exten => s,1,Dial(${ARG2},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-NOANS...

...he line when it rings? I mean, how do I ask it, does it pick up the line or just monitor the "ring" signal and wait for me to pick up a phone then connect me to it? [inbound-analog] exten => s,1,NoOp exten => s,2,Dial(${PHONE1}&${PHONE2},15,Ttm) exten => s,3,Voicemail(u${PHONE1VM}) exten => s,4,Hangup exten => s,102,Voicemail(b${PHONE1VM}) exten => s,103,Hangup

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

...5@from-sip") in new stack -- Called madsen:password@nattedbox/55555@from-sip -- Hungup 'IAX2[207.61.247.201:4569]/1' == No one is available to answer at this time And here are the config files: *** NAT'd Box *** extensions.conf --------------- [globals] PHONE1=Zap/1 PHONE1VM=18924 CALLFILENAME=foo FOO=foo [intern] include => outbound-fwd include => from-sip [outbound-fwd] exten => _7.,1,Dial(IAX2/madsen:password@liveipbox/${EXTEN}@intern) exten => _7.,2,DISA,no-password|intern [from-sip] exten => 18924,1,Dial(${PHONE1},30,t) exten => 18924,2,Voice...

...ar any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten => s,7,Voicemail(u${PHONE1VM}) exten => s,8,Hangup exten => s,107,Voicemail(b${PHONE1VM}) exten => s,108,Hangup Do you see anything wrong with this ? Regards...Martin -- Anything is possible on paper. -- Ron McAfee

...ound-analog channel=> 4 This is in my zaptel.conf fxoks=1 fxsks=1 loadzone=nl defaultzone=nl This is in my extensions.conf: [inbound-analog] exten => s,1,Wait(1) exten => s,2,Dial(${PHONE1}&${PHONE2},25,tm) exten => s,3,Answer exten => s,4,Wait(1) exten => s,5,Voicemail(u${PHONE1VM}) exten => s,6,Hangup exten => s,7,Wait(2) exten => s,8,Voicemail(b${PHONE1VM}) exten => s,9,Hangup

...Goto(from-sip-post,${EXTEN},1) exten => i,1,Hangup exten => h,1,Hangup [from-sip-post] exten => 16463752819,1,Dial(${PHONE1},20,Ttm) exten => 16463752819,2,Playback(transfer) exten => 16463752819,3,Macro(dialiconnect,${MYCELLPHONE},20) exten => 16463752819,4,Voicemail(u${PHONE1VM}) exten => 16463752819,5,Hangup exten => 16463752819,102,Voicemail(b${PHONE1VM}) exten => 16463752819,103,Hangup exten => 222,1,Dial(SIP/dave,30,t) exten => 333,1,Dial(SIP/french1,30,t) ;exten => 444,1,Dial(SIP/6666xxxxxxxxxxxxxx@uuuuuuuu:pppp@213.137.73.140,30,t) exten...

...other PBX. These are the relevant entries in extensions, PHONE1=Zap/1 [macro-extensions] exten => s,1,Dial(${ARG1},20) exten => s,2,Voicemail2(u${ARG2}) exten => s,102,Voicemail2(b${ARG2}) exten => s,103,Hangup [home] include=>tozap exten => 2201,1,Macro(extensions,${PHONE1},${PHONE1VM}) exten => 2202,1,Macro(extensions,${PHONE2},${PHONE2VM}) [tozap] ignorepath => 9 exten => _9XXXXXXX,1,Dial(${LINE1}/9WW${EXTEN:1},90,rt) ; is 9 and a second wait for dialing tone from the legacy PBX central, then the number. exten => _9XXXXXXX,2,Congestion I don't know why yet,...

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten