Hi All
I’ve recently setup Asterisk for the first time. I am running version: 1.4.22.1 . I am not able to access voicemail and I think that the problem is on the extensions.conf as I get the error message “no route to destination”. I’ve setup 2 users( User1-6000 and User2-6001). All users are internal to the network
What I would like to happen is :
Call user 1 and ring for 30 seconds. If there is no answer then forward to his voicemail. After the end of the voicemail play “vm-goodbye” and hang up.
If any of the users call the voicemail (7500) they Should be able to authenticate and listen to their messages
Relevant conf files below
Thanks
[code]voicemail.conf
[Personal_mailbox]
6000 = 12345,User1,tz=european|maxmsg=10
6001 = 12345,User2,tz=european|maxmsg=10
users.conf
[6000]
username = User1
type = friend
transfer = yes
mailbox = 6000
call-limit = 100
fullname =
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = yes
vmsecret = 6000
email =
threewaycalling = no
hasdirectory = no
callwaiting = yes
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
secret = 123456
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm,h263
[6001]
username = User2
type = friend
transfer = yes
mailbox = 6001
call-limit = 100
fullname =
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 6001
hasvoicemail = yes
vmsecret = 6001
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
secret = 123456
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm,h263
sip.conf
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr =
srvlookup = yes
allowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite =
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain =
dtmfmode =
dumphistory = yes
externrefresh = 10
fromdomain =
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize =
jbresyncthreshold =
language = en
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
nat =
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = yes
registerattempts = 0
registertimeout = 20
relaxdtmf = no
rtpholdtimeout =
rtptimeout =
sendrpid = no
sipdebug = yes
subscribecontext =
t1min = 100
t38pt_udptl = no
tos_audio = ef
tos_sip = none
tos_video = ef
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = yes
maxcallbitrate = 384
disallow = all
allow = undefined,ulaw,alaw,gsm
extensions.conf
[Home]
exten = 6000,1,Dial(SIP/User1, 30)
exten = 6000,2,VoiceMail(6000@Personal_mailbox,u)
exten = 6000,3,PlayBack(vm-goodbye)
exten = 6000,4,HangUp()
exten = 6001,1,Dial(SIP/User2, 30)
exten = 6001,2,VoiceMail(6001@Personal_mailbox,u)
exten = 6001,3,PlayBack(vm-goodbye)
exten = 6001,4,HangUp()
exten = 7500,1,Answer(500)
exten = 7500,n,VoiceMailMain(@Personal_mailbox,u)
[/code]
Also if I try to restart the service I am getting the following error:
[/etc/asterisk] # /etc/init.d/asterisk.sh restart
RESTART Asterisk
STOP Asterisk
Cannot read termcap database;
using dumb terminal settings.
START Asterisk
