[asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

[Steve Hickel]

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