[asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...
Steve Hickel
smhickel at hickel.info
Tue May 6 09:50:35 CDT 2008
These are the instructions that I followed. I did managed to get the
fast busy to go away, but the RDNIS simply does not seem to work. These
are the instructions that I followed on this project. I have run out of
time trying to get Call Manager 4.x to talk to Asterisk 1.4.
http://www.voip-info.org/wiki/index.php?page_id=2596#editcomments
These instructions although a good start, simply lack the pictures or
images to set up CCM properly, and because of the coding change from
earlier versions, this just doesn't seem to allow voice mail to work.
I have learned a lot about asterisk, but am frustrated by this
experience.
Thanks Sean for the info about the change of the rdnis command format.
Kind regards,
Steve
On Mon, 2008-05-05 at 23:33 -0400, Steve Hickel wrote:
> 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
> >
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> 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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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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