[Asterisk-Users] question re voicemail

Jess Magnaye jess at arretni.com
Mon Jan 5 14:28:37 MST 2004


Hi,
I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy.  i only get continuous ringback and the following message: 

asterisk*CLI> 
    -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack
    -- Called 5104112978
    -- SIP/5104112978-3f88 is ringing
    -- Nobody picked up in 20000 ms

I wonder if my u<extension> and b<extension> config is correct, mispelled, or something else is missing.  Note that ata to ata via * works, as well as getting to VoicemailMain via extension 1234.    Please help.  My config are found below.  I appreciate your help.


sip.conf
-----------

[6882332]
type=friend
username=6882332
secret=test
host=dynamic
defaultip=172.30.200.27
dtmfmode=rfc2833
mailbox=6882332
callerid = "test1" <6882332>
context=sip

[5104112978]
type=friend
username=5104112978
secret=test
host=dynamic
;canreinvite=no
defaultip=172.30.200.26
dtmfmode=rfc2833
mailbox=5104112978
callerid = "test2" <5104112978>
context=sip

extensions.conf
------------------------
[sip]
;ring
exten => 5104112978,1,Dial(SIP/5104112978,20,tr)
exten => 6882332,1,Dial(SIP/6882332,15,tr)
exten => 1111,1,Dial(SIP/1111,5,tr)

;unanswered
exten => 6882332,102,Voicemail,u6882332
exten => 5104112978,102,Voicemail,u5104112978
exten => 1111,102,Voicemail,u1111

;busy
exten => 6882332,103,Voicemail,b6882332
exten => 5104112978,103,Voicemail,b5104112978
exten => 1111,103,Voicemail,b1111

;get message
exten => 1234,1,VoicemailMain(6882332);
exten => 9999,1,VoicemailMain(1111);

voicemail.conf
---------------------

[default]
6882332 => 6882332,test1,jess at arretni.com
5104112978 => 5104112978,test2, jess at arretni.com
9011 => 9011,Asterisk,jess at arretni.com
1111 => 1111,Nada,jess at arretni.com
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