Hi,

The VoiceMail() application is to leave messages in mailboxes, to enable users to access their voicemails, check the application VoiceMailMain().
core show application VoiceMailMain

Regards,

El 09/06/11 17:57, motty.cruz escribió:
Hello, I'm new to this list. I'm trying to configure my Asterisk to have
user access their email. SO far users can leave voicemail but they can't
access voicemail. As you can see I had sip.conf and extensions.conf below.
Please advice how to access configure extensions.conf to have users access
their voicemail.

Thanks in advance.
-motty

SIP.CONF
[general]
context=default
srvlookup=yes
realm = domain.tld
disallow=all
allow=ulaw
allow=gsm
language=en

register =>  user1:pas...@host.domain.tld/user1
register =>  user2:pas...@shost.domain.tld/user2
register =>  user3:pas...@host.domain.tld/user3

[user1]
type=friend             ; make and recieved calls
callerid="user1"<5001>  ; caller id
secret=passwd        ; password
host=dynamic            ; how to find a client
qualify=yes             ; qualify peer is no more than 2000 ms away
nat=yes                 ; this phone is not natted
host=dynamic            ; this device registers with us
fromuser=user1          ; calling from user
fromdomain=domain.tld   ; calling from domain realm
canreinvite=no          ; allow RTP voice traffic to bypass Asterisk
context=internal        ; the internal context controls what we can do
fromdomain=domain.tld   ; from domain -the domain that the username is from
mailbox = 5001@default  ; avoid-Received SIP subscribe for peer without
mailbox-mess

[user3]
type=friend
callerid="user3"<5003>
secret=passwd
host=dynamic
qualify=yes             ; qualify peer is no more than 2000 ms away
nat=yes                 ; this phone is not natted
host=dynamic            ; this device registers with us
fromuser=user3          ; user id from
fromdomain=domain.tld      ; realm of user
canreinvite=no          ; asterisk by default tries to redirect
context=internal        ; the internal context controls what we can do
fromdomain=domain.tld      ; from domain -the domain that the username is
from
mailbox = 5003@default  ; avoid-Received SIP subscribe for peer without
mailbox-mess
(END)

Extensions.conf

[default]
exten =>  5001,1,Dial(SIP/user1,20)
exten =>  5001,2,VoiceMail(5002@default)
exten =>  5001,4,PlayBack(vm-goodbye)
exten =>  5001,5,Wait(2)
exten =>  5001,6,HangUp()

exten =>  5002,1,Dial(SIP/user2)
exten =>  5002,2,VoiceMail(5002@default)
exten =>  5002,3,PlayBack(vm-goodbye)
exten =>  5002,4,Wait(2)
exten =>  5002,5,HangUp()

exten =>  5003,1,Dial(SIP/user3)
exten =>  5003,2,VoiceMail(5002@default)
exten =>  5003,4,PlayBack(vm-goodbye)
exten =>  5003,5,Wait(2)
exten =>  5003,6,HangUp()

[internal]
include = default
exten =>  5XXX,1,Dial(SIP/${EXTEN})
exten =>  5XXX,2,Voicemail(${CALLERID},u)
exten =>  5XXX,3,Hangup


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
                http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
    http://lists.digium.com/mailman/listinfo/asterisk-users
-----------------------------------------------------------------------------------
Este mensaje y sus anexos son para uso exclusivo de sus destinatarios y puede
contener informacion confidencial y/o privada protegida legalmente. Si usted no es el destinatario, se le notifica que cualquier distribucion o reproduccion de este mensaje, o de cualquiera de sus anexos, esta estrictamente prohibida. Si usted ha recibido este mensaje por error, por favor notifiquenos inmediatamente
y elimine su texto original, incluidos los anexos y destruya cualquier 
reproduccion
del mismo. Las opiniones expresadas en este mensaje son responsabilidad 
exclusiva
de quien las emite y no necesariamente reflejan la posicion de Millenium Phone Center S.A, ni comprometen la responsabilidad institucional por el uso que el destinatario haga de las mismas. -----------------------------------------------------------------------------------

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to