Tim,

Make sure in your sip.conf for your inbound provider the setting for "dtmfmode" 
matches what your provider requires, Asterisk defaults to rfc2833 .

Lonnie


On Aug 23, 2017, at 9:20 AM, Tim Turpin <ttur...@z-harris.com> wrote:

> 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+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 ':
> 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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> engaging tech sites, Slashdot.org! 
> http://sdm.link/slashdot_______________________________________________
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