[asterisk-users] Access Voicemail Asterisk 1.8 FreeBSD 8.2

[Miguel Molina]

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:passwd at host.domain.tld/user1
> register =>  user2:passwd at shost.domain.tld/user2
> register =>  user3:passwd at 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 at 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 at 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 at 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 at 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 at 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
>
>
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