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Re: [Astlinux-users] Question about setting up AstLinux as voicemail server
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From: The C. K. <eld...@ya...> - 2017-08-23 23:51:19
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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...@lo...>
To: AstLinux Users Mailing List <ast...@li...>
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 <tt...@z-...> 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...@ke...]
> 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 <tt...@z-...> 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...@ke...]
> 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 <tt...@z-...> 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...@lo...]
> 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 <tt...@z-...> 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:ast...@li...]
> > 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(${boxnumber}) exten => a,n, Hangup
> >
> >
> >
> > -Christopher
> >
> >
> > From: Tim Turpin <tt...@z-...>
> > To: 'AstLinux Users Mailing List'
> > <ast...@li...>
> > Sent: Wednesday, August 23, 2017 8:14 AM
> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> > voicemail server
> >
> > This appears to possibly work for one mailbox user. We have a couple
> thousand users, all dialing in via DID, and the process needs to be the same
> for all users. My current extensions.conf looks like this:
> >
> > [inbound]
> > exten => _NXXNXXXXXX,1,Answer
> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
> > exten => _NXXNXXXXXX,n,Voicemail(${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...@ke...]
> > 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 <tt...@z-...> 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...@lo...]
> > 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 <tt...@z-...> 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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