[asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...
Steve Hickel
smhickel at hickel.info
Mon May 5 22:33:09 CDT 2008
Sean,
Here is what I changed. Now I have a fast busy...
Steve
[demo]
exten=s,1,Wait(1)
exten=s,n,Answer
exten=s,n,Set(TIMEOUT(digit)=5)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n(restart),BackGround(demo-congrats)
exten=s,n(instruct),BackGround(demo-instruct)
exten=s,n,WaitExten
exten=2,1,BackGround(demo-moreinfo)
exten=2,n,Goto(s,instruct)
exten=3,1,Set(LANGUAGE()=fr)
exten=3,n,Goto(s,restart)
exten=1000,1,Goto(default,s,1)
exten=1234,1,Playback(transfer,skip)
exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten=1235,1,Voicemail(1234,u)
exten=1236,1,Dial(Console/dsp)
exten=1236,n,Voicemail(1234,b)
exten=#,1,Playback(demo-thanks)
exten=#,n,Hangup
exten=t,1,Goto(#,1)
exten=i,1,Playback(invalid)
exten=500,1,Playback(demo-abouttotry)
exten=500,n,Dial(IAX2/guest at misery.digium.com/s at default)
exten=500,n,Playback(demo-nogo)
exten=500,n,Goto(s,6)
exten=600,1,Playback(demo-echotest)
exten=600,n,Echo
exten=600,n,Playback(demo-echodone)
exten=600,n,Goto(s,6)
exten=76245,1,Macro(page,SIP/Grandstream1)
exten=_7XXX,1,Macro(page,SIP/${EXTEN})
exten=7999,1,Set(TIMEOUT(absolute)=60)
exten=7999,2,Page(Local/Grandstream1 at page&Local/Xlite1 at page&Local/1234 at page/n|d)
exten=7777,1,VoicemailMain
exten=7777,n,Goto(s,6)
[general]
static=yes
writeprotect=no
clearglobalvars=no
autofallthrough=yes
priorityjumping=no
[default]
exten=_230XXXX,1,SetCallerID(${EXTEN:3})
exten=_230XXXX,2,Dial(SIP/28888 at sip)
exten=_230XXXX,3,Answer
exten=_230XXXX,4,Wait,1
exten=_230XXXX,5,Hangup
exten=_231XXXX,1,SetCallerID(${EXTEN:3})
exten=_231XXXX,2,Dial(SIP/28889 at sip)
exten=_231XXXX,3,Answer
exten=_231XXXX,4,Wait,1
exten=_231XXXX,5,Hangup
exten=7777,1,VoiceMailMain
[incoming] exten=7777,1,GotoIf($[${CALLERID(rdnis)}]?2:400)
exten=7777,2,MailboxExists(${CALLERID(rdnis)}@default)
exten=7777,3,Congestion
exten=7777,103,Voicemail(su${CALLERID(rdnis)}
exten=7777,104,Playback(vm-goodbye)
exten=7777,105,Hangup
exten=7777,400,VoicemailMain
_____
From: Sean Dennis [mailto:sean at datawhale.com]
To: smhickel at hickel.info, Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com]
Sent: Mon, 05 May 2008 17:58:32 -0400
Subject: Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...
Steve Hickel wrote:
> I have sip set up on Callmanager 4.x. When others call my ext of 2016 on
> ccm after a busy or no answer, asterisk voice mail answers by saying,
> "Mailbox .... password." I want it to put them into my mailbox so they
> can leave a message. Somehow I must be missing something... Please
> help!
>
> I have spent 19 hours easy on trying to figure this one out.
>
> SIP DN is 7777 on CCM
> VOICEMAIL on Asterisk is 7777.
>
> Here is my sip.conf:
>
> [general]
> context=default
> allowoverlap=no
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> allowexternaldomains=yes
> allowexternalinvites=no
> allowguest=yes
> allowsubscribe=no
> allowtransfer=yes
> alwaysauthreject=no
> autodomain=no
> callevents=no
> compactheaders=no
> dumphistory=no
> g726nonstandard=no
> ignoreregexpire=no
> jbenable=no
> jbforce=no
> jblog=no
> maxcallbitrate=384
> maxexpiry=3600
> minexpiry=60
> nat=no
> notifyringing=no
> pedantic=no
> promiscredir=no
> recordhistory=no
> relaxdtmf=no
> rtcachefriends=no
> rtsavesysname=no
> rtupdate=no
> sendrpid=yes
> sipdebug=no
> t1min=100
> t38pt_udptl=no
> [authentication]
>
> [sip]
> type=friend
> context=incoming
> host=172.20.1.57
> ipaddr=172.20.1.57
> allow=ulaw
> allow=alaw
> nat=no
> canreinvite=yes
> qualify=yes
>
> Here is my voicemail.conf
>
> [zonemessages]
> eastern=America/New_York|'vm-received' Q 'digits/at' IMp
> central=America/Chicago|'vm-received' Q 'digits/at' IMp
> central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
> military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
> european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
> [other]
>
> [general]
> format=wav49|gsm|wav
> serveremail=asterisk
> attach=yes
> skipms=3000
> maxsilence=10
> silencethreshold=128
> maxlogins=3
> emaildateformat=%A, %B %d, %Y at %r
> sendvoicemail=yes
> attachfmt=wav
> deletevoicemail=no
> envelope=no
> maxgreet=60
> maxmessage=120
> maxmsg=100
> minmessage=1
> operator=yes
> review=yes
> saycid=no
> sayduration=yes
> mailcmd=/usr/sbin/sendmail -t
> externotify=/var/libasterisk/scripts/vm.sh
> [default]
> 2016=1234,Steve,steve at abc.com
>
> Here is the relevant parts of my extensions.conf:
>
> [macro-dialout-callmanager]
> exten=s,1,ChanIsAvail(SIP/sip)
> exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1)
> exten=s,3,Dial(${AVAILCHAN}/${ARG1})
> exten=s,4,Hangup
> exten=s,102,Congestion
> [incoming]
> exten=7777,1,GotoIf($[${RDNIS}]?2:400)
> exten=7777,2,MailboxExists(${RDNIS}@default
> exten=7777,3,Congestion
> exten=7777,103,Voicemail(su${RDNIS})
> exten=7777,104,Playback(vm-goodbye)
> exten=7777,105,Hangup
> exten=7777,400,VoicemailMain
> [general]
> static=yes
> writeprotect=no
> clearglobalvars=no
> autofallthrough=yes
> priorityjumping=no
> [default]
> exten=_230XXXX,1,SetCallerID(${EXTEN:3})
> exten=_230XXXX,2,Dial(SIP/28888 at 172.20.1.57)
> exten=_230XXXX,3,Answer
> exten=_230XXXX,4,Wait,1
> exten=_230XXXX,5,Hangup
> exten=_231XXXX,1,SetCallerID(${EXTEN:3})
> exten=_231XXXX,2,Dial(SIP/28889 at 172.20.1.57)
> exten=_231XXXX,3,Answer
> exten=_231XXXX,4,Wait,1
> exten=_231XXXX,5,Hangup
>
> I am using users.conf, but don't know how that ties in or whether I even
> need it...???
>
> thanks,
>
> Steve
>
>
>
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You didn't mention what version of asterisk, but if you are using
version 1.4.x, in extensions.conf you need to use:
CALLERID(rdnis) instead of just RDNIS
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