Make it to Astricon this year, I'll buy you a drink and you can take the test there! That's what I did last year. :)
Thanks, --Warren Selby, dCAP On May 5, 2011, at 1:27 PM, Sherwood McGowan <[email protected]> wrote: > Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have > dCAP by your name! I've been at this 6-7 years and still haven't gotten off > my butt and taken the tests :D > > On Thu, May 5, 2011 at 1:20 PM, Warren Selby <[email protected]> wrote: > And Sherwood beats me to the punch again :). > > Thanks, > --Warren Selby, dCAP > > On May 5, 2011, at 1:15 PM, Sherwood McGowan <[email protected]> > wrote: > >> No, the variables are channel specific except for when they're inherited, >> which doesn't affect you here >> >> On Thu, May 5, 2011 at 1:02 PM, satish patel <[email protected]> wrote: >> After google i found something and i tried following. I set variable before >> Dial and its give me proper value in "h" extension but now question is if >> multiple user dial multiple extension then will it overwrite current >> variable value ? >> >> exten => s,1,Set(_CALLED_EXT=${ARG2}) >> exten => s,n,Dial(${ARG2}&iax2/${ARG1},20,t) >> >> From: [email protected] >> To: [email protected] >> Date: Thu, 5 May 2011 17:52:54 +0000 >> >> Subject: Re: [asterisk-users] missed call notification >> >> Could you please tell me how ( Syntax ) and where in macro ? >> >> I am not expert in dialplan variables. I appreciate your help >> >> Date: Thu, 5 May 2011 12:44:19 -0500 >> From: [email protected] >> To: [email protected] >> Subject: Re: [asterisk-users] missed call notification >> >> if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the macro, >> you'd get 's'....do it while you still have the called number as the EXTEN >> >> On Thu, May 5, 2011 at 12:42 PM, satish patel <[email protected]> wrote: >> >> Also check for CANCEL, since this should be the status if the caller >> hangs up before the call is picked up. >> >> But CANCEL is return nothing >> >> >> [macro-stdexten] >> exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t) ; Ring the >> interface, 20 seconds maximum, call screening option (or use P for databased >> call screening) >> >> >> >> exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on >> status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) >> ;exten => s,n,Hangup() >> >> exten => s-CANCEL,1,Verbose(Hangup call) >> >> >> >> >> >> >> CLI >> == Spawn extension (macro-stdexten, s, 1) exited non-zero on >> 'SIP/7527-00000023' in macro 'stdexten' >> == Spawn extension (from-sip, 7516, 1) exited non-zero on >> 'SIP/7527-00000023' >> >> >> >> >> >> Look like its going back to original extension :( I hate macro >> >> >> From: [email protected] >> >> To: [email protected] >> Date: Thu, 5 May 2011 17:15:53 +0000 >> >> Subject: Re: [asterisk-users] missed call notification >> >> You want me to do this in macro-stdexten ? I have following dialplan. I >> have used "h" extension in original context because you can't you "h" inside >> macro right ? >> >> [macro-stdexten] >> exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t) ; Ring the >> interface, 20 seconds maximum, call screening option (or use P for databased >> call screening) >> exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on >> status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) >> exten => s,n,Hangup() >> exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, >> send to voicemail w/ unavail announce >> exten => s-NOANSWER,n,Hangup() >> exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to >> voicemail w/ busy announce >> exten => s-BUSY,n,Hangup() >> exten => s-CONGESTION,1,Voicemail(${ARG1},u) ; Like above, write >> a macro for this case >> exten => s-CONGESTION,n,Hangup() >> exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything >> else as no answer >> exten => a,1,VoicemailMain(${ARG1}) ; If they press *, >> send the user into VoicemailMain >> >> >> [from-sip] >> ...blah...blah.. >> >> exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh "" >> "${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}") >> >> >> >> >> >> > From: [email protected] >> > Date: Thu, 5 May 2011 12:10:09 -0500 >> > To: [email protected] >> > Subject: Re: [asterisk-users] missed call notification >> > >> > Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, >> > then reference that variable in your h exten. >> > >> > Thanks, >> > --Warren Selby, dCAP >> > >> > On May 5, 2011, at 11:59 AM, satish patel <[email protected]> wrote: >> > >> > > Hi All, >> > > >> > > I am using >> > > http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/ >> > > to implement missed call feature. and i modify script to grab email >> > > address from voicemail.conf >> > > >> > > But i am not able to see DEST extension in this script ? what would be >> > > the variable to get destination extension so base on that i can grab >> > > email address of user from voicemail.conf >> > > >> > > exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh "" >> > > "${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}") >> > > >> > > Calling from 7527<--to--->7101 but i can see only 7527 not dest 7101 >> > > >> > > >> > > CLI outout >> > > -- Executing [h@from-sip:1] System("SIP/7527-0000000d", >> > > "/var/lib/asterisk/agi-bin/processcallemail.sh "" "7527" "Guest" >> > > "CANCEL" """) in new stack >> > > shirley*CLI> exit >> > > >> > > -- >> > > _____________________________________________________________________ >> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > > http://www.asterisk.org/hello >> > > >> > > asterisk-users mailing list >> > > To UNSUBSCRIBE or update options visit: >> > > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > -- >> > _____________________________________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > http://www.asterisk.org/hello >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- _____________________________________________________________________ -- >> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >> Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or >> update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> Sherwood McGowan >> Telecommunications and VOIP Consultant >> >> >> -- _____________________________________________________________________ -- >> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >> Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or >> update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> -- _____________________________________________________________________ -- >> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >> Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or >> update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> Sherwood McGowan >> Telecommunications and VOIP Consultant >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Sherwood McGowan > Telecommunications and VOIP Consultant > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
