Check that the * key is not being captured for some other purpose (grep
into other .conf files). Check that you can match the * key outside of
voicemail... use WaitExten() and validate that your dialplan sees that.
You can also go into the asterisk console ("asterisk -r") and turn on
verbose and debug... e.g. "core set verbose 999" and "core set debug 999"
and watch in the console.... make sure that logger.conf has a line that
says "console => notice,warning,error,debug,verbose" else you might not get
the debug and verbose messages into your console.
David
On Wed, Aug 23, 2017 at 10:53 AM, Tim Turpin <ttur...@z-harris.com> wrote:
> If I change my config to direct the call to VoiceMailMain(), I can log in
> with DTMF digits, so I know the carrier is passing tones. And Asterisk is
> recognizing them.
> Thanks.
>
> -----Original Message-----
> From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com]
> Sent: Wednesday, August 23, 2017 10:51 AM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> voicemail server
>
> 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+Boxe
> > s
> >
> > Lonnie
> >
> >
> > ----------------------------------------------------------------------
> > ------
> > --
> > Check out the vibrant tech community on one of the world's most
> > 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.
> >
> >
> > ----------------------------------------------------------------------
> > -------- Check out the vibrant tech community on one of the world's
> > most engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> > _______________________________________________
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> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
> >
> > ----------------------------------------------------------------------
> > -------- Check out the vibrant tech community on one of the world's
> > most 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.
> >
> >
> > ----------------------------------------------------------------------
> > -------- Check out the vibrant tech community on one of the world's
> > most 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.
>
>
> ------------------------------------------------------------
> ----------------
> --
> Check out the vibrant tech community on one of the world's most 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.
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> 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.
>
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