Getting closer, I think.
I’m starting to wonder if the DTMF ‘*’ is being recognized at all. Now the
caller is dropped into the proper mailbox, but pressing ‘*’ does nothing.
Here’s extensions.conf:
[inbound]
exten => _NXXNXXXXXX,1,Answer
exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
exten => _NXXNXXXXXX,n,Set(boxnumber=${EXTEN}) ; set a variable for box
number
exten => _NXXNXXXXXX,n,NoOp(${boxnumber}) ; test for variable
exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})
exten => _NXXNXXXXXX,n,Hangup
exten => a,1,VoiceMailMain(${boxnumber}) ; user dialed * in greeting.
send them to their mailbox
exten => a,n,Hangup
Here’s the response when calling the DID number 9373506524:
Connected to Asterisk 11.25.1 currently running on SST (pid = 415)
== Using SIP RTP CoS mark 5
-- Executing [9373506524@inbound:1] Answer("SIP/voipms-00000037", "") in
new stack
-- Executing [9373506524@inbound:2] NoOp("SIP/voipms-00000037",
"inbound-phone-call") in new stack
-- Executing [9373506524@inbound:3] Set("SIP/voipms-00000037",
"boxnumber=9373506524") in new stack
-- Executing [9373506524@inbound:4] NoOp("SIP/voipms-00000037",
"9373506524") in new stack
-- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000037",
"9373506524") in new stack
-- <SIP/voipms-00000037> Playing
'/var/spool/asterisk/voicemail/default/9373506524/temp.slin' (language 'en')
-- <SIP/voipms-00000037> Playing 'vm-intro.ulaw' (language 'en')
-- <SIP/voipms-00000037> Playing 'beep.ulaw' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/9373506524/tmp/2u8Hzw format: wav,
0x2addfc001798
Is there any setting that would not allow the ‘*’ to be recognized during the
greeting?
From: The Cadillac Kid via Astlinux-users
[mailto:astlinux-users@lists.sourceforge.net]
Sent: Wednesday, August 23, 2017 8:32 AM
To: AstLinux Users Mailing List
Cc: The Cadillac Kid
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
set a variable first... the issue is that ${EXTEN} changes to 'a' when you *
out... ${EXTEN} is the current extension you are workign with and you want to
go to the original dialed extension.
[inbound]
exten => _NXXNXXXXXX,1,Answer
exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
; set a variable for box number
exten => _NXXNXXXXXX,n,Set(boxumber=${EXTEN})
exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})
;exten => _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})
exten = > _NXXNXXXXXX,n,Hangup
; user dialed * in greeting. send them to their mailbox
exten => a, 1, VoicemailMain(${boxnumber})
exten => a,n, Hangup
-Christopher
_____
From: Tim Turpin <ttur...@z-harris.com>
To: 'AstLinux Users Mailing List' <astlinux-users@lists.sourceforge.net>
Sent: Wednesday, August 23, 2017 8:14 AM
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
This appears to possibly work for one mailbox user. We have a couple thousand
users, all dialing in via DID, and the process needs to be the same for all
users. My current extensions.conf looks like this:
[inbound]
exten => _NXXNXXXXXX,1,Answer
exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
exten => _NXXNXXXXXX,n,Voicemail(${EXTEN})
;exten => _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})
;exten => a, 1, VoicemailMain(${EXTEN})
I’ve played with the ‘a’ extension in different formats, but can’t seem to make
it work. In the current configuration, when a caller dials in, it plays the
greeting for that particular mailbox. If I comment out the third line and
un-comment the fourth, the caller drops into their box with the ability to log
in. I can’t figure out how to utilize the ‘a’ extension to allow the user to
press ‘*’ to login while listening to his greeting (the fifth line).
I’m using information about the ‘a’ extension from the following sites:
>From ’ <https://www.voip-info.org/wiki-asterisk+standard+extensions>
>https://www.voip-info.org/wiki-asterisk+standard+extensions1 ':
a: Called when user presses ‘*’ during a voicemail greeting
h: Hangup extension
i: invalid extension
o: Operator extension, used for operator exit by pressing zero in voicemail
s: Start extension in context
t: Timeout extension
T: AbsoluteTimeout() extension
Also, from ’ <https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail>
https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail ':
Also. during the prompt if the caller presses:
’*’ - the call jumps to extension ‘a’ in the current voicemail context.
Example:
Exten => a, 1, VoicemailMain(@default)
Exten => a, 2, Hangup
Being a novice at Asterisk, I have to assume that I’m not following the proper
coding format, or I’m not applying the ‘a’ extension properly. From what I
have read on these two web pages, I think that this is the application to use,
but I’m just not applying it properly.
From: David Kerr [mailto:da...@kerr.net]
Sent: Tuesday, August 22, 2017 5:30 PM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
Tim,
You are going to want to use the Background() app to play the greeting with
the WaitExten() app to wait for a keypress (if they wait til the very end of
the greeting before pressing) and then the Authenticate() app to get a PIN to
proceed to whatever action is permitted. Something like this (untested but
should be close enough)...
[leavemessage]
exten = s,1,NoOp(voicemail)
same = n,Ringing()
same = n,Wait(2)
same = n,Answer()
same = n(start),Set(TIMEOUT(response)=1)
same = n,Set(TIMEOUT(digit)=1)
same = n,Background(record/NoAnswer) ; my custom message, press 1 or wait to
leave a msg
same = n,WaitExten(1)
exten = 1,1,Voicemail(101,us) ; caller pressed 1
same = n,NoOp(Back from voicemail)
same = n,Hangup()
exten = _[*],1,VoiceMailMain(101,sa(0)) ; caller pressed *
same = n,NoOp(Back from voicemailmain)
same = n,Hangup()
exten = t,1,Voicemail(101,us) ; timeout, leave a message. could GoTo(1,1)
same = n,NoOp(Back from voicemail)
same = n,Hangup()
exten = i,1,Playback(pbx-invalid) ; standard invalid key pressed msg.
same = n,Goto(s,start)
exten = h,1,Hangup()
David
On Tue, Aug 22, 2017 at 3:04 PM, Tim Turpin <ttur...@z-harris.com> wrote:
Thank you for the fast reply.
I loaded up the AstLinux last week. I've been able to figure out most of
what I need, except for a way to route incoming DID calls to voicemail,
allowing the caller to be able to press '*' while hearing the mailbox
greeting and then be handed off to 'VoiceMailMain()' to log into their box.
If '*' isn't pressed, the caller would just drop into the mailbox to leave a
message.
It seems like it should be easy to set up, but it's really kicking my butt
right now, and I'm just trying to determine my best avenue for assistance in
figuring this out. I'll try the Asterisk forums and see if they can offer
any help.
Thanks again.
Tim
-----Original Message-----
From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com]
Sent: Tuesday, August 22, 2017 2:31 PM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Question about setting up AstLinux as
voicemail server
On Aug 22, 2017, at 11:49 AM, Tim Turpin <ttur...@z-harris.com> wrote:
> I'm new to the Asterisk world, and I'm trying to use AstLinux to replicate
an existing voicemail environment, and I have several configuration
questions. Is this the proper forum for these questions, or do I send the
questions somewhere else?
>
> Thanks.
> Tim.
Hi Tim,
First, using AstLinux as a dedicated voicemail server, using a small x86
appliance and SSD storage or Virtual Machine Guest is a good approach.
This mailing list is mostly dedicated to AstLinux Project specific
questions, Asterisk voicemail.conf, sip.conf and extensions.conf
configurations are best asked in the Asterisk support groups. If you have
things all but working and have reached a brick wall using AstLinux ... you
can give this list a try.
Keep in mind that using AstLinux, you will be required to generate the base
extensions.conf text file for yourself, AstLinux has a basic web interface
and "Users" tab that can help manage your voicemail users. As a starting
point you might spin-up the "Guest VM x86-64bit (Video Console)" Install ISO
in a virtual machine to give you a playground to test before purchasing any
hardware.
Alternatively, if coding a extensions.conf is not your cup-of-tea you might
query this mailing list for off-line consulting help.
Here is a reference to give you the flavor of the configuration ...
Configuring Voice Mail Boxes
https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes
Lonnie
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_______________________________________________
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Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
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engaging tech sites, Slashdot.org! http://sdm.link/slashdot
_______________________________________________
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.