I think this is the syntax you are looking
for
[sip]
exten =>
5104112978,1,Dial(SIP/5104112978,20,tr)
exten
=> 5104112978,2,Voicemail,u5104112978
exten =>
5104112978,102,Voicemail,b5104112978
----- Original Message -----
Sent: Monday, January 05, 2004 4:28
PM
Subject: [Asterisk-Users] question re
voicemail
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
------------------------
voicemail.conf
---------------------
[default]
6882332 =>
6882332,test1,[EMAIL PROTECTED]
9011 => 9011,Asterisk,[EMAIL PROTECTED]
1111
=> 1111,Nada,[EMAIL PROTECTED]
