[Serusers] FW: SER Asterisk Voicemail

Steve Blair blairs at isc.upenn.edu
Tue Feb 15 13:18:57 CET 2005


Aisling:

   You need a phone number that "connects" to your Asterisk server.
This will become the "lead" number for voicemail. One way is to
purchase an analog line card and plug it into an existing phone jack.
The other way is to dedicate a number which is sent from the PSTN
to your VoIP environment via a gateway. The latter is how I do it.

   In either case when someone calls this "lead" number the call needs
to get to Asterisk. I'll assume you have this much or can build this much
based on your environment.

   The next thing you will need is an entry in the extensions.conf file 
that
sends calls to this lead number to voicemailmain. You may also need
special treatment in sip.conf depending upon your environment.
Here is what I do in extensions.conf to provide a menu of choice for
people calling the "lead" number.

exten => 68000,1,Goto(pennmainmenu,s,1)

[pennmainmenu]
exten => s,1,Ringing                            ; 2 seconds of ringback
exten => s,2,Answer
exten => s,3,Playback(upenn/welcome-2-penn)     ; Play general welcome
exten => s,4,Playback(upenn/welcome-N-listen) ; Play please listen menu 
options may have changed
exten => s,5,Wait,1                             ; Wait a second, just 
for fun
exten => s,6,Background(upenn/press1)           ; Play press 1 to access 
voicemail system
exten => s,7,Background(upenn/press2)           ; Play press 2 for local 
weather report
exten => s,8,Background(upenn/press3)           ; Play press 3 for local 
date and time
exten => s,9,Background(upenn/press4)           ; Play press 4 to access 
the Penn Zoo
exten => s,10,Background(upenn/press5)          ; Play press 5 for echo test
exten => s,11,Background(upenn/press6)          ; Play press 6, ask for 
pin, instruct, record, playack.
;
; Menu option 1
;
exten => 1,1,VoicemailMain                      ; Goto Voice Mail Main menu
exten => 1,2,Hangup                             ; Hang them up.


   Pressing one gets the caller to the exten => 1,1,VoicemailMain statement
which is what gets them to the voicemail maim menu.

-Steve

Aisling O'Driscoll wrote:

>Would someone be so kind as to give me a few guidelines for "dial-in"
>access to asterisk so that the user can access voicemail, based on
>the below email. I think that is what is happening, the caller is
>leaving a message which is being saved on Asterisk buts ince the
>users are registered with ser they cant access the voicemail.
>I already have an account for each user in sip.conf,extensions.conf
>and voicemail.conf on Asterisk as suggested in the archives.
>
>Thanks a million,
>Aisling.
>
>---- Original Message ----
>From: blairs at isc.upenn.edu
>To: ashling.odriscoll at cit.ie
>Subject: Re: [Serusers] FW: SER Asterisk Voicemail
>Date: Mon, 14 Feb 2005 13:22:38 -0500
>
>  
>
>>If the message is only sent as an email attachment 
>>(delete=yes,attach=yes) then
>>the user must listen to it by playing the attached wav file on their
>>pc.
>>
>>If the message is saved on the Asterisk server then you need to
>>provide
>>"dial-in" access to Asterisk that sends the caller to VoiceMailMain. 
>>From there
>>they can access their mailbox and manage messages.
>>
>>_Steve
>>
>>Aisling O'Driscoll wrote:
>>
>>    
>>
>>>Any more ideas on my below mail? If a user is registered with SER
>>>      
>>>
>>and
>>    
>>
>>>leaves a voicemail message with asterisk (by using rewritehostport
>>>etc in ser.cfg), then how is the user supposed to listen to the
>>>message afterwards? Is there any other way other than the MWI
>>>      
>>>
>>method??
>>    
>>
>>>Thnaksm
>>>Aisling.
>>>
>>>---- Original Message ----
>>>From: ashling.odriscoll at cit.ie
>>>To: asterisk-users at lists.digium.com
>>>Subject: FW: SER Asterisk Voicemail
>>>Date: Thu, 10 Feb 2005 16:45:53 -0000
>>>
>>>Hi all,
>>>
>>>I have SER and Asterisk set up together with ser handling user
>>>registrations and asterisk providing voicemail services. When I ring
>>>a phone and it doesnt answer after a designated amount of time, the
>>>request is forwarded to asterisk, and I can leave a message. 
>>>
>>>Now, this may seem a ridiculous question but how can I listen to my
>>>message afterwards? I have read about a solution by Java Rockx using
>>>sipsak for sending mwi sip notify messages to the phone but is there
>>>a simpler way which I am blindly ignoring??
>>>
>>>Thank you in advance,
>>>Aisling.
>>>
>>>
>>>-------------------Legal 
>>>      
>>>
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>>
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>>>
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>>-- 
>> 
>>ISC Network Engineering
>>The University of Pennsylvania
>>3401 Walnut Street, Suite 221A
>>Philadelphia, PA 19104  
>>
>>
>>voice: 215-573-8396 
>>
>>      215-746-8001
>>
>>fax: 215-898-9348    
>>
>>sip:blairs at upenn.edu
>>
>>
>>-------------------Legal 
>>Disclaimer---------------------------------------
>>
>>The above electronic mail transmission is confidential and intended
>>only for the person to whom it is addressed. Its contents may be
>>protected by legal and/or professional privilege. Should it be
>>received by you in error please contact the sender at the above
>>quoted email address. Any unauthorised form of reproduction of this
>>message is strictly prohibited. The Institute does not guarantee the
>>security of any information electronically transmitted and is not
>>liable if the information contained in this communication is not a
>>proper and complete record of the message as transmitted by the
>>sender nor for any delay in its receipt.
>>    
>>
>
>
>-------------------Legal  Disclaimer---------------------------------------
>
>The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
>  
>

-- 
  
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  


voice: 215-573-8396 

       215-746-8001

fax: 215-898-9348    

sip:blairs at upenn.edu




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