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Jess Magnaye

2004-Jan-05 14:28 UTC

[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
------------------------
[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@arretni.com
5104112978 => 5104112978,test2, jess@arretni.com
9011 => 9011,Asterisk,jess@arretni.com
1111 => 1111,Nada,jess@arretni.com
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Brian West

2004-Jan-05 17:47 UTC

head link

[Asterisk-Users] question re voicemail

> 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@arretni.com
> 5104112978 => 5104112978,test2, jess@arretni.com
> 9011 => 9011,Asterisk,jess@arretni.com
> 1111 => 1111,Nada,jess@arretni.com
I would do this:

exten => 1111,1,Dial(SIP/1111,5,tr)
exten => 1111,2,Voicemail,u1111
exten => 1111,102,Voicemail,b1111

Your priority numbering was off a little and its best to group the whole
extension together instead of spreading them out.  Helps to make sure you
don't get lost.  Also that 5 is in seconds.. not rings.

I would do this

[default]
exten => 1111,1,Macro(stdexten|1111)
exten => 6882332,1,Macro(stdexten|6882332)
exten => 5104112978,1,Macro(stdexten|5104112978)


[macro-stdexten]
exten => s,1,Dial(SIP/${ARG1},20,tr)
exten => s,2,Voicemail(u${ARG1})
exten => s,3,Hangup
exten => s,102,Voicemail(b${ARG1})
exten => s,103,Hangup


Hope that helps.


bkw

Sean Cheesman

2004-Jan-05 17:52 UTC

head link

[Asterisk-Users] question re voicemail

Hi Jess,
 
It looks like your problem is with the extension increment.  If there is no
answer in the allotted time, the count increses by one.  If the line is busy,
the count increases by 101.  Also, have you actually created the vm boxes
you're referencing?  Thanks!

Sean

	-----Original Message----- 
	From: Jess Magnaye [mailto:jess@arretni.com] 
	Sent: Mon 1/5/2004 4:28 PM 
	To: asterisk-users@lists.digium.com 
	Cc: 
	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
	------------------------
	[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@arretni.com
	5104112978 => 5104112978,test2, jess@arretni.com
	9011 => 9011,Asterisk,jess@arretni.com
	1111 => 1111,Nada,jess@arretni.com

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Glenn Dalgliesh

2004-Jan-11 18:38 UTC

head link

[Asterisk-Users] question re voicemail

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@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@arretni.com
  5104112978 => 5104112978,test2, jess@arretni.com
  9011 => 9011,Asterisk,jess@arretni.com
  1111 => 1111,Nada,jess@arretni.com
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