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<br><pre>Also check for CANCEL, since this should be the status if the caller<br>hangs up before the call is picked up.<br><br>But CANCEL is return nothing <br><br>[macro-stdexten]<br>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)<br>exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)<br>;exten => s,n,Hangup()<br><br>exten => s-CANCEL,1,Verbose(Hangup call)<br><br><br><br>CLI<br> == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-00000023' in macro 'stdexten'<br> == Spawn extension (from-sip, 7516, 1) exited non-zero on 'SIP/7527-00000023'<br><br><br>Look like its going back to original extension :( I hate macro <br></pre><br><br><hr id="stopSpelling">From: satish_lx@hotmail.com<br>To: asterisk-users@lists.digium.com<br>Date: Thu, 5 May 2011 17:15:53 +0000<br>Subject: Re: [asterisk-users] missed call notification<br><br>
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You want me to do this in macro-stdexten ? I have following dialplan. I have used "h" extension in original context because you can't you "h" inside macro right ? <br><br>[macro-stdexten]<br>exten => s,1,Dial(${ARG2}&iax2/${ARG1},20,t) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)<br>exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)<br>exten => s,n,Hangup()<br>exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce<br>exten => s-NOANSWER,n,Hangup()<br>exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce<br>exten => s-BUSY,n,Hangup()<br>exten => s-CONGESTION,1,Voicemail(${ARG1},u) ; Like above, write a macro for this case<br>exten => s-CONGESTION,n,Hangup()<br>exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer<br>exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain<br><br><br>[from-sip]<br>...blah...blah..<br><br>exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ""
"${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}")<br><br><br><br><br><br>> From: wcselby@selbytech.com<br>> Date: Thu, 5 May 2011 12:10:09 -0500<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] missed call notification<br>> <br>> Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call, then reference that variable in your h exten. <br>> <br>> Thanks,<br>> --Warren Selby, dCAP<br>> <br>> On May 5, 2011, at 11:59 AM, satish patel <satish_lx@hotmail.com> wrote:<br>> <br>> > Hi All,<br>> > <br>> > I am using http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/ to implement missed call feature. and i modify script to grab email address from voicemail.conf <br>> > <br>> > But i am not able to see DEST extension in this script ? what would be the variable to get destination extension so base on that i can grab email address of user from voicemail.conf <br>> > <br>> > exten => h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh "" "${CALLERID(num)}" "${CALLERID(name)}" "${DIALSTATUS}" "${VMSTATUS}")<br>> > <br>> > Calling from 7527<--to--->7101 but i can see only 7527 not dest 7101<br>> > <br>> > <br>> > CLI outout<br>> > -- Executing [h@from-sip:1] System("SIP/7527-0000000d", "/var/lib/asterisk/agi-bin/processcallemail.sh "" "7527" "Guest" "CANCEL" """) in new stack<br>> > shirley*CLI> exit<br>> > <br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello<br>> > <br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>
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