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Sean,<br><br>Here is what I changed. Now I have a fast busy... <br><br>Steve<br><br>&nbsp;[demo]<div style="padding: 2px 2px 3px 3px; background-color: rgb(224, 230, 196); margin-top: 5px; width: 95%; font-size: 9pt; font-family: courier;" context="demo" id="contextContent_demo" align="left">&nbsp;&nbsp;exten=s,1,Wait(1)<br>&nbsp;&nbsp;exten=s,n,Answer<br>&nbsp;&nbsp;exten=s,n,Set(TIMEOUT(digit)=5)<br>&nbsp;&nbsp;exten=s,n,Set(TIMEOUT(response)=10)<br>&nbsp;&nbsp;exten=s,n(restart),BackGround(demo-congrats)<br>&nbsp;&nbsp;exten=s,n(instruct),BackGround(demo-instruct)<br>&nbsp;&nbsp;exten=s,n,WaitExten<br>&nbsp;&nbsp;exten=2,1,BackGround(demo-moreinfo)<br>&nbsp;&nbsp;exten=2,n,Goto(s,instruct)<br>&nbsp;&nbsp;exten=3,1,Set(LANGUAGE()=fr)<br>&nbsp;&nbsp;exten=3,n,Goto(s,restart)<br>&nbsp;&nbsp;exten=1000,1,Goto(default,s,1)<br>&nbsp;&nbsp;exten=1234,1,Playback(transfer,skip)<br>&nbsp;&nbsp;exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})<br>&nbsp;&nbsp;exten=1235,1,Voicemail(1234,u)<br>&nbsp;&nbsp;exten=1236,1,Dial(Console/dsp)<br>&nbsp;&nbsp;exten=1236,n,Voicemail(1234,b)<br>&nbsp;&nbsp;exten=#,1,Playback(demo-thanks)<br>&nbsp;&nbsp;exten=#,n,Hangup<br>&nbsp;&nbsp;exten=t,1,Goto(#,1)<br>&nbsp;&nbsp;exten=i,1,Playback(invalid)<br>&nbsp;&nbsp;exten=500,1,Playback(demo-abouttotry)<br>&nbsp;&nbsp;exten=500,n,Dial(IAX2/guest@misery.digium.com/s@default)<br>&nbsp;&nbsp;exten=500,n,Playback(demo-nogo)<br>&nbsp;&nbsp;exten=500,n,Goto(s,6)<br>&nbsp;&nbsp;exten=600,1,Playback(demo-echotest)<br>&nbsp;&nbsp;exten=600,n,Echo<br>&nbsp;&nbsp;exten=600,n,Playback(demo-echodone)<br>&nbsp;&nbsp;exten=600,n,Goto(s,6)<br>&nbsp;&nbsp;exten=76245,1,Macro(page,SIP/Grandstream1)<br>&nbsp;&nbsp;exten=_7XXX,1,Macro(page,SIP/${EXTEN})<br>&nbsp;&nbsp;exten=7999,1,Set(TIMEOUT(absolute)=60)<br>&nbsp;&nbsp;exten=7999,2,Page(Local/Grandstream1@page&amp;Local/Xlite1@page&amp;Local/1234@page/n|d)<br>&nbsp;&nbsp;exten=7777,1,VoicemailMain<br>&nbsp;&nbsp;exten=7777,n,Goto(s,6)<br></div><div style="padding: 2px 2px 3px 3px; background-color: rgb(77, 84, 35); color: rgb(255, 255, 255); margin-top: 15px; width: 95%; font-family: 'trebuchet ms',helvetica,sans-serif; font-size: 10pt;" context="general" id="context_general" align="left">&nbsp;&nbsp;[general]</div><div style="padding: 2px 2px 3px 3px; background-color: rgb(224, 230, 196); margin-top: 5px; width: 95%; font-size: 9pt; font-family: courier;" context="general" id="contextContent_general" align="left">&nbsp;&nbsp;static=yes<br>&nbsp;&nbsp;writeprotect=no<br>&nbsp;&nbsp;clearglobalvars=no<br>&nbsp;&nbsp;autofallthrough=yes<br>&nbsp;&nbsp;priorityjumping=no<br><div id="div_editcontextContent" style="z-index: 1001; background-color: rgb(224, 230, 196); display: none;" onclick="stopBubble(event)">
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</div></div><div style="padding: 2px 2px 3px 3px; background-color: rgb(77, 84, 35); color: rgb(255, 255, 255); margin-top: 15px; width: 95%; font-family: 'trebuchet ms',helvetica,sans-serif; font-size: 10pt;" context="default" id="context_default" align="left">&nbsp;&nbsp;[default]</div><div style="padding: 2px 2px 3px 3px; background-color: rgb(224, 230, 196); margin-top: 5px; width: 95%; font-size: 9pt; font-family: courier;" context="default" id="contextContent_default" align="left">&nbsp;&nbsp;exten=_230XXXX,1,SetCallerID(${EXTEN:3})<br>&nbsp;&nbsp;exten=_230XXXX,2,Dial(SIP/28888@sip)<br>&nbsp;&nbsp;exten=_230XXXX,3,Answer<br>&nbsp;&nbsp;exten=_230XXXX,4,Wait,1<br>&nbsp;&nbsp;exten=_230XXXX,5,Hangup<br>&nbsp;&nbsp;exten=_231XXXX,1,SetCallerID(${EXTEN:3})<br>&nbsp;&nbsp;exten=_231XXXX,2,Dial(SIP/28889@sip)<br>&nbsp;&nbsp;exten=_231XXXX,3,Answer<br>&nbsp;&nbsp;exten=_231XXXX,4,Wait,1<br>&nbsp;&nbsp;exten=_231XXXX,5,Hangup<br>&nbsp;&nbsp;exten=7777,1,VoiceMailMain<br></div><div style="padding: 2px 2px 3px 3px; background-color: rgb(77, 84, 35); color: rgb(255, 255, 255); margin-top: 15px; width: 95%; font-family: 'trebuchet ms',helvetica,sans-serif; font-size: 10pt;" context="incoming" id="context_incoming" align="left">&nbsp;&nbsp;[incoming]</div>&nbsp;&nbsp;exten=7777,1,GotoIf($[${CALLERID(rdnis)}]?2:400)<br>&nbsp;&nbsp;exten=7777,2,MailboxExists(${CALLERID(rdnis)}@default)<br>&nbsp;&nbsp;exten=7777,3,Congestion<br>&nbsp;&nbsp;exten=7777,103,Voicemail(su${CALLERID(rdnis)}<br>&nbsp;&nbsp;exten=7777,104,Playback(vm-goodbye)<br>&nbsp;&nbsp;exten=7777,105,Hangup<br>&nbsp;&nbsp;exten=7777,400,VoicemailMain<br><blockquote style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px; margin-right: 0px;"><hr><b>From:</b> Sean Dennis [mailto:sean@datawhale.com]<br><b>To:</b> smhickel@hickel.info, Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com]<br><b>Sent:</b> Mon, 05 May 2008 17:58:32 -0400<br><b>Subject:</b> Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working        ...<br><br>Steve Hickel wrote:<br>
&gt; I have sip set up on Callmanager 4.x. When others call my ext of 2016 on<br>
&gt; ccm after a busy or no answer, asterisk voice mail answers by saying,<br>
&gt; "Mailbox .... password." I want it to put them into my mailbox so they<br>
&gt; can leave a message. Somehow I must be missing something... Please<br>
&gt; help! <br>
&gt;<br>
&gt; I have spent 19 hours easy on trying to figure this one out. <br>
&gt;<br>
&gt; SIP DN is 7777 on CCM <br>
&gt; VOICEMAIL on Asterisk is 7777. <br>
&gt;<br>
&gt; Here is my sip.conf: <br>
&gt;<br>
&gt; [general] <br>
&gt; context=default <br>
&gt; allowoverlap=no <br>
&gt; bindport=5060 <br>
&gt; bindaddr=0.0.0.0 <br>
&gt; srvlookup=yes <br>
&gt; allowexternaldomains=yes <br>
&gt; allowexternalinvites=no <br>
&gt; allowguest=yes <br>
&gt; allowsubscribe=no <br>
&gt; allowtransfer=yes <br>
&gt; alwaysauthreject=no <br>
&gt; autodomain=no <br>
&gt; callevents=no <br>
&gt; compactheaders=no <br>
&gt; dumphistory=no <br>
&gt; g726nonstandard=no <br>
&gt; ignoreregexpire=no <br>
&gt; jbenable=no <br>
&gt; jbforce=no <br>
&gt; jblog=no <br>
&gt; maxcallbitrate=384 <br>
&gt; maxexpiry=3600 <br>
&gt; minexpiry=60 <br>
&gt; nat=no <br>
&gt; notifyringing=no <br>
&gt; pedantic=no <br>
&gt; promiscredir=no <br>
&gt; recordhistory=no <br>
&gt; relaxdtmf=no <br>
&gt; rtcachefriends=no <br>
&gt; rtsavesysname=no <br>
&gt; rtupdate=no <br>
&gt; sendrpid=yes <br>
&gt; sipdebug=no <br>
&gt; t1min=100 <br>
&gt; t38pt_udptl=no <br>
&gt; [authentication] <br>
&gt;<br>
&gt; [sip] <br>
&gt; type=friend <br>
&gt; context=incoming <br>
&gt; host=172.20.1.57 <br>
&gt; ipaddr=172.20.1.57 <br>
&gt; allow=ulaw <br>
&gt; allow=alaw <br>
&gt; nat=no <br>
&gt; canreinvite=yes <br>
&gt; qualify=yes <br>
&gt;<br>
&gt; Here is my voicemail.conf <br>
&gt;<br>
&gt; [zonemessages] <br>
&gt; eastern=America/New_York|'vm-received' Q 'digits/at' IMp <br>
&gt; central=America/Chicago|'vm-received' Q 'digits/at' IMp <br>
&gt; central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' <br>
&gt; military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' <br>
&gt; european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM <br>
&gt; [other] <br>
&gt;<br>
&gt; [general] <br>
&gt; format=wav49|gsm|wav <br>
&gt; serveremail=asterisk <br>
&gt; attach=yes <br>
&gt; skipms=3000 <br>
&gt; maxsilence=10 <br>
&gt; silencethreshold=128 <br>
&gt; maxlogins=3 <br>
&gt; emaildateformat=%A, %B %d, %Y at %r <br>
&gt; sendvoicemail=yes <br>
&gt; attachfmt=wav <br>
&gt; deletevoicemail=no <br>
&gt; envelope=no <br>
&gt; maxgreet=60 <br>
&gt; maxmessage=120 <br>
&gt; maxmsg=100 <br>
&gt; minmessage=1 <br>
&gt; operator=yes <br>
&gt; review=yes <br>
&gt; saycid=no <br>
&gt; sayduration=yes <br>
&gt; mailcmd=/usr/sbin/sendmail -t <br>
&gt; externotify=/var/libasterisk/scripts/vm.sh <br>
&gt; [default] <br>
&gt; 2016=1234,Steve,<a href="mailto:steve@abc.com">steve@abc.com</a> <br>
&gt;<br>
&gt; Here is the relevant parts of my extensions.conf: <br>
&gt;<br>
&gt; [macro-dialout-callmanager] <br>
&gt; exten=s,1,ChanIsAvail(SIP/sip) <br>
&gt; exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1) <br>
&gt; exten=s,3,Dial(${AVAILCHAN}/${ARG1}) <br>
&gt; exten=s,4,Hangup <br>
&gt; exten=s,102,Congestion <br>
&gt; [incoming] <br>
&gt; exten=7777,1,GotoIf($[${RDNIS}]?2:400) <br>
&gt; exten=7777,2,MailboxExists(${RDNIS}@default <br>
&gt; exten=7777,3,Congestion <br>
&gt; exten=7777,103,Voicemail(su${RDNIS}) <br>
&gt; exten=7777,104,Playback(vm-goodbye) <br>
&gt; exten=7777,105,Hangup <br>
&gt; exten=7777,400,VoicemailMain <br>
&gt; [general] <br>
&gt; static=yes <br>
&gt; writeprotect=no <br>
&gt; clearglobalvars=no <br>
&gt; autofallthrough=yes <br>
&gt; priorityjumping=no <br>
&gt; [default] <br>
&gt; exten=_230XXXX,1,SetCallerID(${EXTEN:3}) <br>
&gt; exten=_230XXXX,2,Dial(SIP/<a href="mailto:28888@172.20.1.57">28888@172.20.1.57</a>) <br>
&gt; exten=_230XXXX,3,Answer <br>
&gt; exten=_230XXXX,4,Wait,1 <br>
&gt; exten=_230XXXX,5,Hangup <br>
&gt; exten=_231XXXX,1,SetCallerID(${EXTEN:3}) <br>
&gt; exten=_231XXXX,2,Dial(SIP/<a href="mailto:28889@172.20.1.57">28889@172.20.1.57</a>) <br>
&gt; exten=_231XXXX,3,Answer <br>
&gt; exten=_231XXXX,4,Wait,1 <br>
&gt; exten=_231XXXX,5,Hangup <br>
&gt;<br>
&gt; I am using users.conf, but don't know how that ties in or whether I even<br>
&gt; need it...??? <br>
&gt;<br>
&gt; thanks, <br>
&gt;<br>
&gt; Steve<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
&gt;<br>
&gt; asterisk-users mailing list<br>
&gt; To UNSUBSCRIBE or update options visit:<br>
&gt;    <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
&gt;   <br>
<br>
You didn't mention what version of asterisk, but if you are using <br>
version 1.4.x, in extensions.conf you need to use:<br>
<br>
CALLERID(rdnis) instead of just RDNIS<br>
<br>
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