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
