smhickel at hickel.info
Guest
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 Posted: Tue May 06, 2008 9:50 am Post subject: [asterisk-users] Call manager using Asterisk as voicemail se |
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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:
| Quote: | 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:
| Quote: | I have sip set up on Callmanager 4.x. When others call my
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ext of 2016 on
| Quote: | ccm after a busy or no answer, asterisk voice mail answers
|
by saying,
| Quote: | "Mailbox .... password." I want it to put them into my
|
mailbox so they
| Quote: | can leave a message. Somehow I must be missing something...
|
Please
| Quote: | help!
I have spent 19 hours easy on trying to figure this one
|
out.
| Quote: |
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'
| Quote: | military=Zulu|'vm-received' q 'digits/at' H N 'hours'
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'phonetic/z_p'
| Quote: | european=Europe/Copenhagen|'vm-received' a d b 'digits/at'
|
HM
| Quote: | [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
| Quote: | 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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