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(a)isc.upenn.edu
To: ashling.odriscoll(a)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(a)cit.ie
To: asterisk-users(a)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.
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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@upenn.edu