astlinux-users — AstLinux Users Mailing List

[Astlinux-users] Can't log into user interface astlinux

[Roberto R. <rri...@gm...>]

> Hi all,
> I was using astlinux then got a msg that I had no memory so it stopped everything. I rebooted and everything was fine....it happened again but now I can't get access to the site for the past day it so....I can make outbound calls and was getting ready to work on getting inbound working when I noticed the problem any suggestion would be greatly appreciated...ip address has not changed and I can't ssh in either.
> Thanks

Sent from my iPhone

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Lonnie A. <li...@lo...>]

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
> >

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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 <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: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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> _______________________________________________
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> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ----------------------------------------------------------------------
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> _______________________________________________
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>
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pa...@kr....
>
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Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[David K. <da...@ke...>]

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
> >
> >
> > ----------------------------------------------------------------------
> > ------
> > --
> > 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> > pa...@kr....
> >
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
> >
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ----------------
> --
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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 <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: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
>
>
> ----------------------------------------------------------------------
> ------
> --
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....
>
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....
>
>
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


----------------------------------------------------------------------------
--
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


------------------------------------------------------------------------------
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

 

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[David K. <da...@ke...>]

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
> >
> >
> > ----------------------------------------------------------------------
> > ------
> > --
> > 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> > pa...@kr....
> >
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
> >
> >
> > ----------------------------------------------------------------------
> > -------- 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
> > Ast...@li...
> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> > Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ----------------
> --
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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
> 
> 
> ----------------------------------------------------------------------
> ------
> --
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> pa...@kr....
> 
> 
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....
>  
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....
>  
> 
> ----------------------------------------------------------------------
> -------- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


----------------------------------------------------------------------------
--
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Lonnie A. <li...@lo...>]

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+Boxes
> 
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
> 
> 
> ------------------------------------------------------------------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>  
> ------------------------------------------------------------------------------
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> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
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> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>  
> 
> ------------------------------------------------------------------------------
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Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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+extensions> 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...@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+Boxes

Lonnie


----------------------------------------------------------------------------
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tech sites, Slashdot.org! http://sdm.link/slashdot
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


------------------------------------------------------------------------------
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

 

------------------------------------------------------------------------------
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engaging tech sites, Slashdot.org! http://sdm.link/slashdot

_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

 

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[David K. <da...@ke...>]

Also, be aware that voip-info has a lot of pretty outdated information as
it has not been updated for more recent versions of Asterisk.  So while
google is great at finding info and you often land at voip-info, I
recommend also cross referencing to the master Asterisk Wiki at
https://wiki.asterisk.org/wiki/display/AST/Home for specific
features/functions.

And you should avoid repeating exten => _NXXNXXXXXX,n,xxxx on every line,
that is also old notation and prone to accidental errors.  Use same = n,xxx
instead.

David

On Wed, Aug 23, 2017 at 8:31 AM, The Cadillac Kid via Astlinux-users <
ast...@li...> wrote:

> 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
> <https://www.voip-info.org/wiki-asterisk+standard+extensions> ':
>
>
>
>
>
>
> *a: Called when user presses ‘*’ during a voicemail greetingh: Hangup
> extensioni: invalid extensiono: Operator extension, used for operator exit
> by pressing zero in voicemails: Start extension in contextt: Timeout
> extensionT: 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+Boxes
>
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[The C. K. <eld...@ya...>]

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 numberexten => _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
   
#yiv9586204792 #yiv9586204792 -- _filtered #yiv9586204792 {font-family:Helvetica;panose-1:2 11 6 4 2 2 2 2 2 4;} _filtered #yiv9586204792 {panose-1:2 4 5 3 5 4 6 3 2 4;} _filtered #yiv9586204792 {font-family:Calibri;panose-1:2 15 5 2 2 2 4 3 2 4;} _filtered #yiv9586204792 {font-family:Tahoma;panose-1:2 11 6 4 3 5 4 4 2 4;} _filtered #yiv9586204792 {font-family:Verdana;panose-1:2 11 6 4 3 5 4 4 2 4;}#yiv9586204792 #yiv9586204792 p.yiv9586204792MsoNormal, #yiv9586204792 li.yiv9586204792MsoNormal, #yiv9586204792 div.yiv9586204792MsoNormal {margin:0in;margin-bottom:.0001pt;font-size:12.0pt;}#yiv9586204792 a:link, #yiv9586204792 span.yiv9586204792MsoHyperlink {color:blue;text-decoration:underline;}#yiv9586204792 a:visited, #yiv9586204792 span.yiv9586204792MsoHyperlinkFollowed {color:purple;text-decoration:underline;}#yiv9586204792 p {margin-right:0in;margin-left:0in;font-size:12.0pt;}#yiv9586204792 p.yiv9586204792MsoAcetate, #yiv9586204792 li.yiv9586204792MsoAcetate, #yiv9586204792 div.yiv9586204792MsoAcetate {margin:0in;margin-bottom:.0001pt;font-size:8.0pt;}#yiv9586204792 span.yiv9586204792BalloonTextChar {}#yiv9586204792 span.yiv9586204792EmailStyle19 {color:#1F497D;}#yiv9586204792 span.yiv9586204792mention {}#yiv9586204792 span.yiv9586204792badge {}#yiv9586204792 .yiv9586204792MsoChpDefault {} _filtered #yiv9586204792 {margin:1.0in 1.0in 1.0in 1.0in;}#yiv9586204792 div.yiv9586204792WordSection1 {}#yiv9586204792 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() extensionAlso, 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, HangupBeing 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+Boxes

Lonnie


----------------------------------------------------------------------------
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_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


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Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....  ------------------------------------------------------------------------------
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Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

   

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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+extensions> 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...@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+Boxes

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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


------------------------------------------------------------------------------
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

 

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

Thanks, I’ll give a try.

 

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+Boxes

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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....


------------------------------------------------------------------------------
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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

 

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[David K. <da...@ke...>]

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+Boxes
>
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>
>
> ------------------------------------------------------------
> ------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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+Boxes

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
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Lonnie A. <li...@lo...>]

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+Boxes

Lonnie

[Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

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.

[Astlinux-users] Legacy Installer ISO's for Old Hardware

[Lonnie A. <li...@lo...>]

Greetings,

With the official release of AstLinux 1.3.0, Adrian asked a great question about old non-PAE hardware.

We added a WiKi documentation entry for this topic ...

Legacy Installer ISO's for Old Hardware
https://doc.astlinux-project.org/userdoc:legacy-installer-iso

Lonnie

Re: [Astlinux-users] Announcing AstLinux Release: 1.3.0

[Adrian H. <ad...@tt...>]

On Thursday 17 Aug 2017 17:57:27 Lonnie Abelbeck wrote:
> On Aug 17, 2017, at 4:54 PM, Michael Keuter <li...@mk...> wrote:
> >> Am 17.08.2017 um 23:27 schrieb Adrian Hodgson
> >> <ad...@tt...>:
> >> 
> >> Can I just ask?
> >> 
> >> when I check with my old system it says I am using runnix-0.4-8057 and
> >> when I check or ask it to check for updates it says it is running the
> >> latest version.
> >> 
> >> Now I appreciate this is on an old HP thin client HP-T5700 so wondering
> >> if it recognises runnix-0.4-8057 is the latest that will work with the
> >> hardware?
> >> 
> >> 
> >> I have just done the update and I am running astlinux-1.3.0
> >> 
> >> Cheers
> >> 
> >> Adrian
> > 
> > The new 1.3.0 upgrade-script checks if your CPU has PAE abilties.
> > If not, you won't get a runnix-0.5 upgrade. So yes, this will probably be
> > the latest runnix-version for your box.
> > 
> > Michael Keuter
> > mk solutions
> 
> Exactly, I did some checking and the HP thin client HP-T5700 has a Transmeta
> Crusoe CPU which does not have PAE.
> 
> While runnix-0.4-8057 (bootloader) is the last for that hardware, that
> should not matter much going forward.
> 
> BTW, the AstLinux 1.2.10 Installer ISO is the last to boot on non-PAE CPU's,
> so if you want to do a clean ISO install to a non-PAE CPU, use the 1.2.10
> ISO ...
> 
> AstLinux 1.2.10 Generic x86-32bit (Video Console):
> https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.2.
> 10-geni586.iso
> 
> AstLinux 1.2.10 Generic x86-32bit (Serial Console):
> https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.2.
> 10-geni586-serial.iso
> 
> After initially installed with 1.2.10 ...  "System Firmware Upgrade:" to the
> latest ... 1.3+
> 
> Lonnie
> 
> 
Thank you for that information Lonnie.

I will download the iso's and keep safe, it is only  for home use so cope well 
with what I use it for.

Adrian

Re: [Astlinux-users] Announcing AstLinux Release: 1.3.0

[Lonnie A. <li...@lo...>]

On Aug 17, 2017, at 4:54 PM, Michael Keuter <li...@mk...> wrote:

> 
>> Am 17.08.2017 um 23:27 schrieb Adrian Hodgson <ad...@tt...>:
>> 
>> Can I just ask?
>> 
>> when I check with my old system it says I am using runnix-0.4-8057 and when I 
>> check or ask it to check for updates it says it is running the latest version.
>> 
>> Now I appreciate this is on an old HP thin client HP-T5700 so wondering if it 
>> recognises runnix-0.4-8057 is the latest that will work with the hardware?
>> 
>> 
>> I have just done the update and I am running astlinux-1.3.0
>> 
>> Cheers
>> 
>> Adrian
> 
> The new 1.3.0 upgrade-script checks if your CPU has PAE abilties. 
> If not, you won't get a runnix-0.5 upgrade. So yes, this will probably be the latest runnix-version for your box.

> Michael Keuter
> mk solutions

Exactly, I did some checking and the HP thin client HP-T5700 has a Transmeta Crusoe CPU which does not have PAE.

While runnix-0.4-8057 (bootloader) is the last for that hardware, that should not matter much going forward.

BTW, the AstLinux 1.2.10 Installer ISO is the last to boot on non-PAE CPU's, so if you want to do a clean ISO install to a non-PAE CPU, use the 1.2.10 ISO ...

AstLinux 1.2.10 Generic x86-32bit (Video Console):
https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.2.10-geni586.iso

AstLinux 1.2.10 Generic x86-32bit (Serial Console):
https://s3.amazonaws.com/mirror.astlinux-project/downloads/iso/astlinux-1.2.10-geni586-serial.iso

After initially installed with 1.2.10 ...  "System Firmware Upgrade:" to the latest ... 1.3+

Lonnie

Re: [Astlinux-users] Announcing AstLinux Release: 1.3.0

[Adrian H. <ad...@tt...>]

On Thursday 17 Aug 2017 23:54:30 Michael Keuter wrote:
> > Am 17.08.2017 um 23:27 schrieb Adrian Hodgson
> > <ad...@tt...>:
> > 
> > Can I just ask?
> > 
> > when I check with my old system it says I am using runnix-0.4-8057 and
> > when I check or ask it to check for updates it says it is running the
> > latest version.
> > 
> > Now I appreciate this is on an old HP thin client HP-T5700 so wondering if
> > it recognises runnix-0.4-8057 is the latest that will work with the
> > hardware?
> > 
> > 
> > I have just done the update and I am running astlinux-1.3.0
> > 
> > Cheers
> > 
> > Adrian
> 
> The new 1.3.0 upgrade-script checks if your CPU has PAE abilties.
> If not, you won't get a runnix-0.5 upgrade. So yes, this will probably be
> the latest runnix-version for your box.
> > On Thursday 17 Aug 2017 10:01:21 Lonnie Abelbeck wrote:
> >> Announcing AstLinux Release: 1.3.0
> >> 
OK, thank you for the information. :)

Adrian

Re: [Astlinux-users] Announcing AstLinux Release: 1.3.0

[Michael K. <li...@mk...>]

> Am 17.08.2017 um 23:27 schrieb Adrian Hodgson <ad...@tt...>:
> 
> Can I just ask?
> 
> when I check with my old system it says I am using runnix-0.4-8057 and when I 
> check or ask it to check for updates it says it is running the latest version.
> 
> Now I appreciate this is on an old HP thin client HP-T5700 so wondering if it 
> recognises runnix-0.4-8057 is the latest that will work with the hardware?
> 
> 
> I have just done the update and I am running astlinux-1.3.0
> 
> Cheers
> 
> Adrian

The new 1.3.0 upgrade-script checks if your CPU has PAE abilties. 
If not, you won't get a runnix-0.5 upgrade. So yes, this will probably be the latest runnix-version for your box.

> On Thursday 17 Aug 2017 10:01:21 Lonnie Abelbeck wrote:
>> Announcing AstLinux Release: 1.3.0
>> 
>> More Info: AstLinux Project
>> http://www.astlinux-project.org/
>> 
>> AstLinux 1.3.0 Highlights:
>> • Major upgrade to Linux Kernel 3.16.44, including the RUNNIX bootloader
>> • The default serial baud rate is now 115200 instead of the previous 19200
>> • New firewall "net-prefix-translation" plugin, provides NPTv6 (Network
>> Prefix Translation) for IPv6 • Updated firewall "traffic-shaper" plugin,
>> use fq_codel (Fair Queueing CoDel) for both 'htb' and 'hfsc' types • New
>> command "acme-client" to generate Let's Encrypt certificates using the ACME
>> protocol • Added support for VMware Tools, vmw_pvscsi and virtio-scsi disk
>> drivers (genx86_64-vm) • Web Interface enhancements and package upgrades
>> providing important security and bug fixes
>> 
>> Full ChangeLog:
>> http://svn.code.sf.net/p/astlinux/code/tags/1.3.0/docs/ChangeLog.txt
>> 
>> All users are encouraged to upgrade.
>> 
>> AstLinux Team
>> 
>> 
>> ----------------------------------------------------------------------------
>> -- 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
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>> 
>> Donations to support AstLinux are graciously accepted via PayPal to
>> pa...@kr....
> 
> 
> ------------------------------------------------------------------------------
> 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....


Freundliche Grüße

Michael Keuter

-- 
Michael Keuter
mk solutions
August-Bebel-Straße 65
21029 Hamburg
Tel. 040-41493636
Fax 040-41493637

Email: mailto:mk...@mk...
Web:   http://www.mksolutions.info


Michael

http://www.mksolutions.info

Re: [Astlinux-users] Announcing AstLinux Release: 1.3.0

[Adrian H. <ad...@tt...>]

Can I just ask?

when I check with my old system it says I am using runnix-0.4-8057 and when I 
check or ask it to check for updates it says it is running the latest version.

Now I appreciate this is on an old HP thin client HP-T5700 so wondering if it 
recognises runnix-0.4-8057 is the latest that will work with the hardware?


I have just done the update and I am running astlinux-1.3.0

Cheers

Adrian


On Thursday 17 Aug 2017 10:01:21 Lonnie Abelbeck wrote:
> Announcing AstLinux Release: 1.3.0
> 
> More Info: AstLinux Project
> http://www.astlinux-project.org/
> 
> AstLinux 1.3.0 Highlights:
> • Major upgrade to Linux Kernel 3.16.44, including the RUNNIX bootloader
> • The default serial baud rate is now 115200 instead of the previous 19200
> • New firewall "net-prefix-translation" plugin, provides NPTv6 (Network
> Prefix Translation) for IPv6 • Updated firewall "traffic-shaper" plugin,
> use fq_codel (Fair Queueing CoDel) for both 'htb' and 'hfsc' types • New
> command "acme-client" to generate Let's Encrypt certificates using the ACME
> protocol • Added support for VMware Tools, vmw_pvscsi and virtio-scsi disk
> drivers (genx86_64-vm) • Web Interface enhancements and package upgrades
> providing important security and bug fixes
> 
> Full ChangeLog:
> http://svn.code.sf.net/p/astlinux/code/tags/1.3.0/docs/ChangeLog.txt
> 
> All users are encouraged to upgrade.
> 
> AstLinux Team
> 
> 
> ----------------------------------------------------------------------------
> -- 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
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....

[Astlinux-users] AstLinux development has moved to GitHub

[Lonnie A. <li...@lo...>]

The AstLinux Project is now hosted on GitHub, both 'svn checkout' and 'git clone' are supported: 

AstLinux Project at GitHub
https://github.com/astlinux-project/astlinux

All current development is on GitHub. Pull Requests are welcomed. 

Note: The Sourgeforge mailing lists [astlinux-users] and [astlinux-devel] will continue to be used as before.

For AstLinux users, there should be no noticeable change.

For AstLinux developers, custom builds will require a new checkout:
--
svn co https://github.com/astlinux-project/astlinux.git/trunk astlinux/trunk
--
The Developer docs have been updated, but not much has changed:
https://doc.astlinux-project.org/devdoc:documentation


Special thanks to David Kerr for assisting in researching and testing, making the migration straightforward.

AstLinux Team

[Astlinux-users] Announcing AstLinux Release: 1.3.0

[Lonnie A. <li...@lo...>]

Announcing AstLinux Release: 1.3.0

More Info: AstLinux Project
http://www.astlinux-project.org/

AstLinux 1.3.0 Highlights:
• Major upgrade to Linux Kernel 3.16.44, including the RUNNIX bootloader
• The default serial baud rate is now 115200 instead of the previous 19200
• New firewall "net-prefix-translation" plugin, provides NPTv6 (Network Prefix Translation) for IPv6
• Updated firewall "traffic-shaper" plugin, use fq_codel (Fair Queueing CoDel) for both 'htb' and 'hfsc' types
• New command "acme-client" to generate Let's Encrypt certificates using the ACME protocol
• Added support for VMware Tools, vmw_pvscsi and virtio-scsi disk drivers (genx86_64-vm)
• Web Interface enhancements and package upgrades providing important security and bug fixes

Full ChangeLog:
http://svn.code.sf.net/p/astlinux/code/tags/1.3.0/docs/ChangeLog.txt

All users are encouraged to upgrade.

AstLinux Team