Found the problem! In voicemail.conf, there is a setting called ‘exitcontext’
It was set to ‘vm-operator’. I found this on the voip-info.org site:
exitcontext
Optional context to drop the user into after he/she has pressed * or 0 to exit
voicemail. If not set, pressing * or 0 will return the caller to the last
context they were in before being sent to voicemail (assuming that context has
a 'a' or 'o' extension).
After commenting out this setting, pressing ‘ * ‘ will drop me right into the
proper mailbox asking for a password.
Thanks again for all your help and ideas on this.
From: The Cadillac Kid via Astlinux-users
[mailto:astlinux-users@lists.sourceforge.net]
Sent: Wednesday, August 23, 2017 7:51 PM
To: AstLinux Users Mailing List
Cc: The Cadillac Kid
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
it definitely works while playing the greeting...
-- Executing [s@macro-ringphone:207] CELGenUserEvent("SIP/4999-00000001",
"VMSCOVER,1503518585.1,4001") in new stack
-- Executing [s@macro-ringphone:208] VoiceMail("SIP/4999-00000001",
"4001@default,u") in new stack
> 0x9af5428 -- Probation passed - setting RTP source address to
172.16.37.1:50478
-- <SIP/4999-00000001> Playing 'vm-theperson.ulaw' (language 'en')
-- <SIP/4999-00000001> Playing 'digits/4.ulaw' (language 'en')
-- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4214 __ast_read: DTMF
begin '*' received on SIP/4999-00000001
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4218 __ast_read: DTMF
begin ignored '*' on SIP/4999-00000001
-- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4128 __ast_read: DTMF
end '*' received on SIP/4999-00000001, duration 160 ms
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4198 __ast_read: DTMF
end passthrough '*' on SIP/4999-00000001
-- Executing [a@macro-ringphone:1] Set("SIP/4999-00000001",
"mailboxnum=4001") in new stack
-- Executing [a@macro-ringphone:2] NoOp("SIP/4999-00000001", "called
by:featureset-dial") in new stack
-- Executing [a@macro-ringphone:3] Set("SIP/4999-00000001", "boxpass=7623")
in new stack
-- Executing [a@macro-ringphone:4] Set("SIP/4999-00000001",
"adminpro=admin-") in new stack
-- Executing [a@macro-ringphone:5] GotoIf("SIP/4999-00000001",
"0?voicemenu-checkvm,s,logmeout") in new stack
-- Executing [a@macro-ringphone:6] Goto("SIP/4999-00000001",
"voicemail-login,s,starlog") in new stack
-- Goto (voicemail-login,s,7)
make sure that your a extension is recognized by asterisk.. do a dialplan show
of your context.. below is an example of mine where the 'a' extension shows..
I looked through the source code of 11.20 and didnt see nay config options that
need set to enable it.. be sure you did a dialplan reload after you make
changes to your contexts (or are you running realtime?)
-Christopher
VM*CLI> dialplan show macro-ringphone
[ Context 'macro-ringphone' created by 'pbx_config' ]
'a' => 1. Set(mailboxnum=${cidreceiver}) [pbx_config]
2. NoOp(called by:${MACRO_CONTEXT}) [pbx_config]
3. Set(boxpass=${DB(vmpass/${mailboxnum})}) [pbx_config]
4. Set(adminpro=${IF($[$["${DB(active/${mailboxnum})}" !=
"yes"] & $["${DB(active/${mailboxnum})}" != "no"]]?admin-)}) [pbx_config]
5. GotoIf($[$["${mailboxnum}" = "${boxpass}"] ||
["${boxpass}" = "${DEFAULT_VM_PASSCODE}"]]?voicemenu-checkvm,s,logmeout)
[pbx_config]
6. Goto(voicemail-login,s,starlog) [pbx_config]
7. Hangup() [pbx_config]
_____
From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net>
Sent: Wednesday, August 23, 2017 4:32 PM
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
server
Tim,
For testing you might try also adding the 'd' option to VoiceMail()
--
d - Accept digits for a new extension in context c, if played during the
greeting. Context defaults to the current context.
--
try "1" first then "*" .
https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail
>From reading the docs I'm not sure if
--
* - Jump to the 'a' extension in the current dialplan context.
--
works while playing the greeting.
Lonnie
On Aug 23, 2017, at 3:09 PM, Tim Turpin <ttur...@z-harris.com> wrote:
> I pressed ‘*’ twice while listening to my unavailable greeting, nothing
> happened.
>
> I believe Asterisk is doing nothing with the ‘*’:
>
>
> -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000046",
> "9373506524,u") in new stack
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: app_voicemail.c:6413
> leave_voicemail: Before find_user
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set
> channel SIP/voipms-00000046 to write format slin
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3446
> ast_rtp_write: Ooh, format changed from unknown to ulaw
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3481
> ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3343
> ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x2addf4026628'
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer
> ticks per second
> -- <SIP/voipms-00000046> Playing
> '/var/spool/asterisk/voicemail/default/9373506524/unavail.slin' (language
> 'en')
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4333
> ast_rtp_read: 0x2addf402b830 -- Probation learning mode pass with source
> address 72.9.246.170:13730
> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:38] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:4285 __sip_autodestruct: Auto
> destroying SIP dialog '0190242c0812028377b2281e2df47b3b@72.9.246.170:5060'
> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:6379 sip_pvt_dtor: Destroying SIP
> dialog 0190242c0812028377b2281e2df47b3b@72.9.246.170:5060
> [Aug 23 15:50:39] DEBUG[482]: rtp_engine.c:226 instance_destructor: Destroyed
> RTP instance '0x2addf4002d98'
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (58 requested / 58 actual) timer
> ticks per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks
> per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks
> per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks
> per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set
> channel SIP/voipms-00000046 to write format ulaw
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer
> ticks per second
>
> -- <SIP/voipms-00000046> Playing 'vm-intro.ulaw' (language 'en')
> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591
> create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
>
> [Aug 23 15:50:42] DEBUG[482]: chan_sip.c:9057 find_call: = Looking for Call
> ID: 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 (Checking From) --From
> tag as2b5c0e97 --To-tag as7ac59689
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:28533 handle_incoming:
> **** Received BYE (8) - Command in SIP BYE
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:138
> ast_sockaddr_split_hostport: Splitting '72.9.246.170:5060' into...
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:192
> ast_sockaddr_split_hostport: ...host '72.9.246.170' and port '5060'.
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3387 sip_alreadygone:
> Setting SIP_ALREADYGONE on dialog
> 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: res_rtp_asterisk.c:4755
> ast_rtp_remote_address_set: Setting RTCP address on RTP instance
> '0x2addf4026628'
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:29442
> stop_session_timer: Session timer stopped: 1 -
> 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:27149
> handle_request_bye: Received bye, issuing owner hangup
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3731 __sip_xmit: Trying
> to put 'SIP/2.0 200' onto UDP socket destined for 72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:6789 __ast_pbx_run: Spawn
> extension (inbound,9373506524,5) exited non-zero on 'SIP/voipms-00000046'
> == Spawn extension (inbound, 9373506524, 5) exited non-zero on
> 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2662
> ast_softhangup_nolock: Soft-Hanging up channel 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:2111 new_find_extension:
> return at end of func
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks
> per second
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595
> ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks
> per second
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2841 ast_hangup: Hanging
> up channel 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: chan_sip.c:6929 sip_hangup: Hangup
> call SIP/voipms-00000046, SIP callid
> 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4755
> ast_rtp_remote_address_set: Setting RTCP address on RTP instance
> '0x2addf4026628'
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:345 _ast_device_state: No
> provider found, checking channel drivers for SIP - voipms
> [Aug 23 15:50:42] DEBUG[438]: chan_sip.c:29982 sip_devicestate: Checking
> device state for peer voipms
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:477 do_state_change: Changing
> state for SIP/voipms - state 1 (Not in use)
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:452 devstate_event: device
> 'SIP/voipms' state '1'
> [Aug 23 15:50:42] DEBUG[509]: app_queue.c:1924 handle_statechange: Device
> 'SIP/voipms' changed to state '1' (Not in use) but we don't care because
> they're not a member of any queue.
>
> It doesn’t appear to be taking any action at all. The system continues to
> record the message and delivers out to email. Is it possible that the ‘a’
> extension is broken?
>
>
> From: David Kerr [mailto:da...@kerr.net]
> Sent: Wednesday, August 23, 2017 2:00 PM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail
> server
>
> That tells you that Asterisk is detecting the tone. Doesn't tell you what it
> is doing with it... so you still need to trace dialplan execution (turn off
> debug, leave verbose on) to see what action it is taking on the tone.
>
> David
>
> On Wed, Aug 23, 2017 at 12:13 PM, Tim Turpin <ttur...@z-harris.com> wrote:
> 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@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
> > <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+Boxe
> > s
> >
> > Lonnie
> >
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