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Sean,<br><br>Here is what I changed. Now I have a fast busy... <br><br>Steve<br><br> [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"> exten=s,1,Wait(1)<br> exten=s,n,Answer<br> exten=s,n,Set(TIMEOUT(digit)=5)<br> exten=s,n,Set(TIMEOUT(response)=10)<br> exten=s,n(restart),BackGround(demo-congrats)<br> exten=s,n(instruct),BackGround(demo-instruct)<br> exten=s,n,WaitExten<br> exten=2,1,BackGround(demo-moreinfo)<br> exten=2,n,Goto(s,instruct)<br> exten=3,1,Set(LANGUAGE()=fr)<br> exten=3,n,Goto(s,restart)<br> exten=1000,1,Goto(default,s,1)<br> exten=1234,1,Playback(transfer,skip)<br> exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})<br> exten=1235,1,Voicemail(1234,u)<br> exten=1236,1,Dial(Console/dsp)<br> exten=1236,n,Voicemail(1234,b)<br> exten=#,1,Playback(demo-thanks)<br> exten=#,n,Hangup<br> exten=t,1,Goto(#,1)<br> exten=i,1,Playback(invalid)<br> exten=500,1,Playback(demo-abouttotry)<br> exten=500,n,Dial(IAX2/guest@misery.digium.com/s@default)<br> exten=500,n,Playback(demo-nogo)<br> exten=500,n,Goto(s,6)<br> exten=600,1,Playback(demo-echotest)<br> exten=600,n,Echo<br> exten=600,n,Playback(demo-echodone)<br> exten=600,n,Goto(s,6)<br> exten=76245,1,Macro(page,SIP/Grandstream1)<br> exten=_7XXX,1,Macro(page,SIP/${EXTEN})<br> exten=7999,1,Set(TIMEOUT(absolute)=60)<br> exten=7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)<br> exten=7777,1,VoicemailMain<br> 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"> [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"> static=yes<br> writeprotect=no<br> clearglobalvars=no<br> autofallthrough=yes<br> 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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                <input id="save_contextContent" value="Save" class="input8" onclick="update_contextContent();" type="button">
                <input id="cancel_contextContent" value="Cancel" class="input8" onclick="cancel_contextContent();" type="button">
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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"> [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"> exten=_230XXXX,1,SetCallerID(${EXTEN:3})<br> exten=_230XXXX,2,Dial(SIP/28888@sip)<br> exten=_230XXXX,3,Answer<br> exten=_230XXXX,4,Wait,1<br> exten=_230XXXX,5,Hangup<br> exten=_231XXXX,1,SetCallerID(${EXTEN:3})<br> exten=_231XXXX,2,Dial(SIP/28889@sip)<br> exten=_231XXXX,3,Answer<br> exten=_231XXXX,4,Wait,1<br> exten=_231XXXX,5,Hangup<br> 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"> [incoming]</div> exten=7777,1,GotoIf($[${CALLERID(rdnis)}]?2:400)<br> exten=7777,2,MailboxExists(${CALLERID(rdnis)}@default)<br> exten=7777,3,Congestion<br> exten=7777,103,Voicemail(su${CALLERID(rdnis)}<br> exten=7777,104,Playback(vm-goodbye)<br> exten=7777,105,Hangup<br> 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>
> I have sip set up on Callmanager 4.x. When others call my ext of 2016 on<br>
> ccm after a busy or no answer, asterisk voice mail answers by saying,<br>
> "Mailbox .... password." I want it to put them into my mailbox so they<br>
> can leave a message. Somehow I must be missing something... Please<br>
> help! <br>
><br>
> I have spent 19 hours easy on trying to figure this one out. <br>
><br>
> SIP DN is 7777 on CCM <br>
> VOICEMAIL on Asterisk is 7777. <br>
><br>
> Here is my sip.conf: <br>
><br>
> [general] <br>
> context=default <br>
> allowoverlap=no <br>
> bindport=5060 <br>
> bindaddr=0.0.0.0 <br>
> srvlookup=yes <br>
> allowexternaldomains=yes <br>
> allowexternalinvites=no <br>
> allowguest=yes <br>
> allowsubscribe=no <br>
> allowtransfer=yes <br>
> alwaysauthreject=no <br>
> autodomain=no <br>
> callevents=no <br>
> compactheaders=no <br>
> dumphistory=no <br>
> g726nonstandard=no <br>
> ignoreregexpire=no <br>
> jbenable=no <br>
> jbforce=no <br>
> jblog=no <br>
> maxcallbitrate=384 <br>
> maxexpiry=3600 <br>
> minexpiry=60 <br>
> nat=no <br>
> notifyringing=no <br>
> pedantic=no <br>
> promiscredir=no <br>
> recordhistory=no <br>
> relaxdtmf=no <br>
> rtcachefriends=no <br>
> rtsavesysname=no <br>
> rtupdate=no <br>
> sendrpid=yes <br>
> sipdebug=no <br>
> t1min=100 <br>
> t38pt_udptl=no <br>
> [authentication] <br>
><br>
> [sip] <br>
> type=friend <br>
> context=incoming <br>
> host=172.20.1.57 <br>
> ipaddr=172.20.1.57 <br>
> allow=ulaw <br>
> allow=alaw <br>
> nat=no <br>
> canreinvite=yes <br>
> qualify=yes <br>
><br>
> Here is my voicemail.conf <br>
><br>
> [zonemessages] <br>
> eastern=America/New_York|'vm-received' Q 'digits/at' IMp <br>
> central=America/Chicago|'vm-received' Q 'digits/at' IMp <br>
> central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' <br>
> military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' <br>
> european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM <br>
> [other] <br>
><br>
> [general] <br>
> format=wav49|gsm|wav <br>
> serveremail=asterisk <br>
> attach=yes <br>
> skipms=3000 <br>
> maxsilence=10 <br>
> silencethreshold=128 <br>
> maxlogins=3 <br>
> emaildateformat=%A, %B %d, %Y at %r <br>
> sendvoicemail=yes <br>
> attachfmt=wav <br>
> deletevoicemail=no <br>
> envelope=no <br>
> maxgreet=60 <br>
> maxmessage=120 <br>
> maxmsg=100 <br>
> minmessage=1 <br>
> operator=yes <br>
> review=yes <br>
> saycid=no <br>
> sayduration=yes <br>
> mailcmd=/usr/sbin/sendmail -t <br>
> externotify=/var/libasterisk/scripts/vm.sh <br>
> [default] <br>
> 2016=1234,Steve,<a href="mailto:steve@abc.com">steve@abc.com</a> <br>
><br>
> Here is the relevant parts of my extensions.conf: <br>
><br>
> [macro-dialout-callmanager] <br>
> exten=s,1,ChanIsAvail(SIP/sip) <br>
> exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1) <br>
> exten=s,3,Dial(${AVAILCHAN}/${ARG1}) <br>
> exten=s,4,Hangup <br>
> exten=s,102,Congestion <br>
> [incoming] <br>
> exten=7777,1,GotoIf($[${RDNIS}]?2:400) <br>
> exten=7777,2,MailboxExists(${RDNIS}@default <br>
> exten=7777,3,Congestion <br>
> exten=7777,103,Voicemail(su${RDNIS}) <br>
> exten=7777,104,Playback(vm-goodbye) <br>
> exten=7777,105,Hangup <br>
> exten=7777,400,VoicemailMain <br>
> [general] <br>
> static=yes <br>
> writeprotect=no <br>
> clearglobalvars=no <br>
> autofallthrough=yes <br>
> priorityjumping=no <br>
> [default] <br>
> exten=_230XXXX,1,SetCallerID(${EXTEN:3}) <br>
> exten=_230XXXX,2,Dial(SIP/<a href="mailto:28888@172.20.1.57">28888@172.20.1.57</a>) <br>
> exten=_230XXXX,3,Answer <br>
> exten=_230XXXX,4,Wait,1 <br>
> exten=_230XXXX,5,Hangup <br>
> exten=_231XXXX,1,SetCallerID(${EXTEN:3}) <br>
> exten=_231XXXX,2,Dial(SIP/<a href="mailto:28889@172.20.1.57">28889@172.20.1.57</a>) <br>
> exten=_231XXXX,3,Answer <br>
> exten=_231XXXX,4,Wait,1 <br>
> exten=_231XXXX,5,Hangup <br>
><br>
> I am using users.conf, but don't know how that ties in or whether I even<br>
> need it...??? <br>
><br>
> thanks, <br>
><br>
> Steve<br>
><br>
><br>
><br>
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> <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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