I won’t copy in the entire session (way too much info), but here’s the result
of my pressing *,*,1,2,3,4,5,6,#. It looks as though Asterisk is seeing the
DTMF.
[Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772
[Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772
[Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772
[Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772
[Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 49 (1), at 72.9.246.170:12772
[Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 49 (1), at 72.9.246.170:12772
[Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 72.9.246.170:12772
[Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 72.9.246.170:12772
[Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 51 (3), at 72.9.246.170:12772
[Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 51 (3), at 72.9.246.170:12772
[Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 52 (4), at 72.9.246.170:12772
[Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 52 (4), at 72.9.246.170:12772
[Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 53 (5), at 72.9.246.170:12772
[Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 53 (5), at 72.9.246.170:12772
[Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 72.9.246.170:12772
[Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 72.9.246.170:12772
[Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at 72.9.246.170:12772
[Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591
create_dtmf_frame: Creating END DTMF Frame: 35 (#), at 72.9.246.170:12772
From: David Kerr [mailto:da...@kerr.net]
Sent: Wednesday, August 23, 2017 11:05 AM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
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 <tel:%5B9373506524> @inbound:1]
> Answer("SIP/voipms-00000037", "")
in new stack
> -- Executing [9373506524 <tel:%5B9373506524> @inbound:2]
> NoOp("SIP/voipms-00000037",
"inbound-phone-call") in new stack
> -- Executing [9373506524 <tel:%5B9373506524> @inbound:3]
> Set("SIP/voipms-00000037",
"boxnumber=9373506524") in new stack
> -- Executing [9373506524 <tel:%5B9373506524> @inbound:4]
> NoOp("SIP/voipms-00000037",
"9373506524") in new stack
> -- Executing [9373506524 <tel:%5B9373506524> @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
>
>
> ----------------------------------------------------------------------
> ------
> --
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> Astlinux-users mailing list
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>
> 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
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pay...@krisk.org.
>
> ----------------------------------------------------------------------
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> 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
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pay...@krisk.org.
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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.
------------------------------------------------------------------------------
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engaging tech sites, Slashdot.org! http://sdm.link/slashdot
_______________________________________________
Astlinux-users mailing list
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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.
------------------------------------------------------------------------------
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engaging tech sites, Slashdot.org! http://sdm.link/slashdot
_______________________________________________
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Donations to support AstLinux are graciously accepted via PayPal to
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