[Asterisk-Users] question re voicemail
Glenn Dalgliesh
asterisk at techhat.com
Sun Jan 11 18:38:52 MST 2004
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 -----
From: Jess Magnaye
To: asterisk-users at lists.digium.com
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,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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