Tim, For testing you might try also adding the 'd' option to VoiceMail() -- d - Accept digits for a new extension in context c, if played during the greeting. Context defaults to the current context. -- try "1" first then "*" .
https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail >From reading the docs I'm not sure if -- * - Jump to the 'a' extension in the current dialplan context. -- works while playing the greeting. Lonnie On Aug 23, 2017, at 3:09 PM, Tim Turpin <ttur...@z-harris.com> wrote: > I pressed ‘*’ twice while listening to my unavailable greeting, nothing > happened. > > I believe Asterisk is doing nothing with the ‘*’: > > > -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000046", > "9373506524,u") in new stack > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: app_voicemail.c:6413 > leave_voicemail: Before find_user > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set > channel SIP/voipms-00000046 to write format slin > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3446 > ast_rtp_write: Ooh, format changed from unknown to ulaw > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3481 > ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160 > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3343 > ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x2addf4026628' > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer > ticks per second > -- <SIP/voipms-00000046> Playing > '/var/spool/asterisk/voicemail/default/9373506524/unavail.slin' (language > 'en') > [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4333 > ast_rtp_read: 0x2addf402b830 -- Probation learning mode pass with source > address 72.9.246.170:13730 > [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730 > [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730 > [Aug 23 15:50:38] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730 > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730 > [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:4285 __sip_autodestruct: Auto > destroying SIP dialog '0190242c0812028377b2281e2df47b3b@72.9.246.170:5060' > [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:6379 sip_pvt_dtor: Destroying SIP > dialog 0190242c0812028377b2281e2df47b3b@72.9.246.170:5060 > [Aug 23 15:50:39] DEBUG[482]: rtp_engine.c:226 instance_destructor: Destroyed > RTP instance '0x2addf4002d98' > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (58 requested / 58 actual) timer > ticks per second > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks > per second > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks > per second > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks > per second > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set > channel SIP/voipms-00000046 to write format ulaw > [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer > ticks per second > > -- <SIP/voipms-00000046> Playing 'vm-intro.ulaw' (language 'en') > [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730 > [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730 > > [Aug 23 15:50:42] DEBUG[482]: chan_sip.c:9057 find_call: = Looking for Call > ID: 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 (Checking From) --From > tag as2b5c0e97 --To-tag as7ac59689 > [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:28533 handle_incoming: > **** Received BYE (8) - Command in SIP BYE > [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:138 > ast_sockaddr_split_hostport: Splitting '72.9.246.170:5060' into... > [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:192 > ast_sockaddr_split_hostport: ...host '72.9.246.170' and port '5060'. > [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3387 sip_alreadygone: > Setting SIP_ALREADYGONE on dialog > 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 > [Aug 23 15:50:42] DEBUG[482][C-00000052]: res_rtp_asterisk.c:4755 > ast_rtp_remote_address_set: Setting RTCP address on RTP instance > '0x2addf4026628' > [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:29442 > stop_session_timer: Session timer stopped: 1 - > 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 > [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:27149 > handle_request_bye: Received bye, issuing owner hangup > [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3731 __sip_xmit: Trying > to put 'SIP/2.0 200' onto UDP socket destined for 72.9.246.170:5060 > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:6789 __ast_pbx_run: Spawn > extension (inbound,9373506524,5) exited non-zero on 'SIP/voipms-00000046' > == Spawn extension (inbound, 9373506524, 5) exited non-zero on > 'SIP/voipms-00000046' > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2662 > ast_softhangup_nolock: Soft-Hanging up channel 'SIP/voipms-00000046' > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:2111 new_find_extension: > return at end of func > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks > per second > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 > ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks > per second > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2841 ast_hangup: Hanging > up channel 'SIP/voipms-00000046' > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: chan_sip.c:6929 sip_hangup: Hangup > call SIP/voipms-00000046, SIP callid > 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 > [Aug 23 15:50:42] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4755 > ast_rtp_remote_address_set: Setting RTCP address on RTP instance > '0x2addf4026628' > [Aug 23 15:50:42] DEBUG[438]: devicestate.c:345 _ast_device_state: No > provider found, checking channel drivers for SIP - voipms > [Aug 23 15:50:42] DEBUG[438]: chan_sip.c:29982 sip_devicestate: Checking > device state for peer voipms > [Aug 23 15:50:42] DEBUG[438]: devicestate.c:477 do_state_change: Changing > state for SIP/voipms - state 1 (Not in use) > [Aug 23 15:50:42] DEBUG[438]: devicestate.c:452 devstate_event: device > 'SIP/voipms' state '1' > [Aug 23 15:50:42] DEBUG[509]: app_queue.c:1924 handle_statechange: Device > 'SIP/voipms' changed to state '1' (Not in use) but we don't care because > they're not a member of any queue. > > It doesn’t appear to be taking any action at all. The system continues to > record the message and delivers out to email. Is it possible that the ‘a’ > extension is broken? > > > From: David Kerr [mailto:da...@kerr.net] > Sent: Wednesday, August 23, 2017 2:00 PM > To: AstLinux Users Mailing List > Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail > server > > That tells you that Asterisk is detecting the tone. Doesn't tell you what it > is doing with it... so you still need to trace dialplan execution (turn off > debug, leave verbose on) to see what action it is taking on the tone. > > David > > On Wed, Aug 23, 2017 at 12:13 PM, Tim Turpin <ttur...@z-harris.com> wrote: > I won’t copy in the entire session (way too much info), but here’s the result > of my pressing *,*,1,2,3,4,5,6,#. It looks as though Asterisk is seeing > the DTMF. > > > [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772 > > [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772 > > [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772 > > [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772 > > [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 49 (1), at 72.9.246.170:12772 > > [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 49 (1), at 72.9.246.170:12772 > > [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 72.9.246.170:12772 > > [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 72.9.246.170:12772 > > [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 51 (3), at 72.9.246.170:12772 > > [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 51 (3), at 72.9.246.170:12772 > > [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 52 (4), at 72.9.246.170:12772 > > [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 52 (4), at 72.9.246.170:12772 > > [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 53 (5), at 72.9.246.170:12772 > > [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 53 (5), at 72.9.246.170:12772 > > [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 72.9.246.170:12772 > > [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 72.9.246.170:12772 > > [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at 72.9.246.170:12772 > > [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 > create_dtmf_frame: Creating END DTMF Frame: 35 (#), at 72.9.246.170:12772 > > > > > From: David Kerr [mailto:da...@kerr.net] > Sent: Wednesday, August 23, 2017 11:05 AM > To: AstLinux Users Mailing List > Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail > server > > Check that the * key is not being captured for some other purpose (grep into > other .conf files). Check that you can match the * key outside of > voicemail... use WaitExten() and validate that your dialplan sees that. You > can also go into the asterisk console ("asterisk -r") and turn on verbose and > debug... e.g. "core set verbose 999" and "core set debug 999" and watch in > the console.... make sure that logger.conf has a line that says "console => > notice,warning,error,debug,verbose" else you might not get the debug and > verbose messages into your console. > > David > > On Wed, Aug 23, 2017 at 10:53 AM, Tim Turpin <ttur...@z-harris.com> wrote: > If I change my config to direct the call to VoiceMailMain(), I can log in > with DTMF digits, so I know the carrier is passing tones. And Asterisk is > recognizing them. > Thanks. > > -----Original Message----- > From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com] > Sent: Wednesday, August 23, 2017 10:51 AM > To: AstLinux Users Mailing List > Subject: Re: [Astlinux-users] Question about setting up AstLinux as > voicemail server > > Tim, > > Make sure in your sip.conf for your inbound provider the setting for > "dtmfmode" matches what your provider requires, Asterisk defaults to rfc2833 > . > > Lonnie > > > On Aug 23, 2017, at 9:20 AM, Tim Turpin <ttur...@z-harris.com> wrote: > > > Getting closer, I think. > > > > I'm starting to wonder if the DTMF '*' is being recognized at all. Now > the caller is dropped into the proper mailbox, but pressing '*' does > nothing. > > Here's extensions.conf: > > > > [inbound] > > > > exten => _NXXNXXXXXX,1,Answer > > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call) > > exten => _NXXNXXXXXX,n,Set(boxnumber=${EXTEN}) ; set a variable for box > number > > exten => _NXXNXXXXXX,n,NoOp(${boxnumber}) ; test for variable > > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber}) > > exten => _NXXNXXXXXX,n,Hangup > > exten => a,1,VoiceMailMain(${boxnumber}) ; user dialed * in > greeting. send them to their mailbox > > exten => a,n,Hangup > > > > > > Here's the response when calling the DID number 9373506524: > > > > Connected to Asterisk 11.25.1 currently running on SST (pid = 415) > > == Using SIP RTP CoS mark 5 > > -- Executing [9373506524@inbound:1] Answer("SIP/voipms-00000037", "") > in new stack > > -- Executing [9373506524@inbound:2] NoOp("SIP/voipms-00000037", > "inbound-phone-call") in new stack > > -- Executing [9373506524@inbound:3] Set("SIP/voipms-00000037", > "boxnumber=9373506524") in new stack > > -- Executing [9373506524@inbound:4] NoOp("SIP/voipms-00000037", > "9373506524") in new stack > > -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000037", > "9373506524") in new stack > > -- <SIP/voipms-00000037> Playing > '/var/spool/asterisk/voicemail/default/9373506524/temp.slin' (language 'en') > > -- <SIP/voipms-00000037> Playing 'vm-intro.ulaw' (language 'en') > > -- <SIP/voipms-00000037> Playing 'beep.ulaw' (language 'en') > > -- Recording the message > > -- x=0, open writing: > > /var/spool/asterisk/voicemail/default/9373506524/tmp/2u8Hzw format: > > wav, 0x2addfc001798 > > > > Is there any setting that would not allow the '*' to be recognized during > the greeting? > > > > > > > > > > From: The Cadillac Kid via Astlinux-users > > [mailto:astlinux-users@lists.sourceforge.net] > > Sent: Wednesday, August 23, 2017 8:32 AM > > To: AstLinux Users Mailing List > > Cc: The Cadillac Kid > > Subject: Re: [Astlinux-users] Question about setting up AstLinux as > > voicemail server > > > > set a variable first... the issue is that ${EXTEN} changes to 'a' when you > * out... ${EXTEN} is the current extension you are workign with and you > want to go to the original dialed extension. > > > > [inbound] > > exten => _NXXNXXXXXX,1,Answer > > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call) > > ; set a variable for box number > > exten => _NXXNXXXXXX,n,Set(boxumber=${EXTEN}) > > > > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber}) > > ;exten => _NXXNXXXXXX,n,VoiceMailMain(${EXTEN}) > > exten = > _NXXNXXXXXX,n,Hangup > > > > ; user dialed * in greeting. send them to their mailbox > > > > exten => a, 1, VoicemailMain(${boxnumber}) exten => a,n, Hangup > > > > > > > > -Christopher > > > > > > From: Tim Turpin <ttur...@z-harris.com> > > To: 'AstLinux Users Mailing List' > > <astlinux-users@lists.sourceforge.net> > > Sent: Wednesday, August 23, 2017 8:14 AM > > Subject: Re: [Astlinux-users] Question about setting up AstLinux as > > voicemail server > > > > This appears to possibly work for one mailbox user. We have a couple > thousand users, all dialing in via DID, and the process needs to be the same > for all users. My current extensions.conf looks like this: > > > > [inbound] > > exten => _NXXNXXXXXX,1,Answer > > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call) > > exten => _NXXNXXXXXX,n,Voicemail(${EXTEN}) ;exten => > > _NXXNXXXXXX,n,VoiceMailMain(${EXTEN}) > > ;exten => a, 1, VoicemailMain(${EXTEN}) > > > > I've played with the 'a' extension in different formats, but can't seem to > make it work. In the current configuration, when a caller dials in, it > plays the greeting for that particular mailbox. If I comment out the third > line and un-comment the fourth, the caller drops into their box with the > ability to log in. I can't figure out how to utilize the 'a' extension to > allow the user to press '*' to login while listening to his greeting (the > fifth line). > > > > I'm using information about the 'a' extension from the following sites: > > > > From ' https://www.voip-info.org/wiki-asterisk+standard+extensions1 ': > > a: Called when user presses '*' during a voicemail greeting > > h: Hangup extension > > i: invalid extension > > o: Operator extension, used for operator exit by pressing zero in > > voicemail > > s: Start extension in context > > t: Timeout extension > > T: AbsoluteTimeout() extension > > Also, from ' https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail ': > > Also. during the prompt if the caller presses: > > '*' - the call jumps to extension 'a' in the current voicemail context. > > Example: > > Exten => a, 1, VoicemailMain(@default) Exten => a, 2, Hangup Being a > > novice at Asterisk, I have to assume that I'm not following the proper > coding format, or I'm not applying the 'a' extension properly. From what I > have read on these two web pages, I think that this is the application to > use, but I'm just not applying it properly. > > > > From: David Kerr [mailto:da...@kerr.net] > > Sent: Tuesday, August 22, 2017 5:30 PM > > To: AstLinux Users Mailing List > > Subject: Re: [Astlinux-users] Question about setting up AstLinux as > > voicemail server > > > > Tim, > > You are going to want to use the Background() app to play the greeting > with the WaitExten() app to wait for a keypress (if they wait til the very > end of the greeting before pressing) and then the Authenticate() app to get > a PIN to proceed to whatever action is permitted. Something like this > (untested but should be close enough)... > > > > [leavemessage] > > exten = s,1,NoOp(voicemail) > > same = n,Ringing() > > same = n,Wait(2) > > same = n,Answer() > > same = n(start),Set(TIMEOUT(response)=1) same = > > n,Set(TIMEOUT(digit)=1) same = n,Background(record/NoAnswer) ; my > > custom message, press 1 or wait to leave a msg same = n,WaitExten(1) > > exten = 1,1,Voicemail(101,us) ; caller pressed 1 same = n,NoOp(Back > > from voicemail) same = n,Hangup() exten = > > _[*],1,VoiceMailMain(101,sa(0)) ; caller pressed * same = n,NoOp(Back > > from voicemailmain) same = n,Hangup() exten = t,1,Voicemail(101,us) ; > > timeout, leave a message. could GoTo(1,1) same = n,NoOp(Back from > > voicemail) same = n,Hangup() exten = i,1,Playback(pbx-invalid) ; > > standard invalid key pressed msg. > > same = n,Goto(s,start) > > exten = h,1,Hangup() > > > > David > > > > > > > > On Tue, Aug 22, 2017 at 3:04 PM, Tim Turpin <ttur...@z-harris.com> wrote: > > Thank you for the fast reply. > > > > I loaded up the AstLinux last week. I've been able to figure out most > > of what I need, except for a way to route incoming DID calls to > > voicemail, allowing the caller to be able to press '*' while hearing > > the mailbox greeting and then be handed off to 'VoiceMailMain()' to log > into their box. > > If '*' isn't pressed, the caller would just drop into the mailbox to > > leave a message. > > > > It seems like it should be easy to set up, but it's really kicking my > > butt right now, and I'm just trying to determine my best avenue for > > assistance in figuring this out. I'll try the Asterisk forums and see > > if they can offer any help. > > > > Thanks again. > > > > Tim > > > > > > > > -----Original Message----- > > From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com] > > Sent: Tuesday, August 22, 2017 2:31 PM > > To: AstLinux Users Mailing List > > Subject: Re: [Astlinux-users] Question about setting up AstLinux as > > voicemail server > > > > > > On Aug 22, 2017, at 11:49 AM, Tim Turpin <ttur...@z-harris.com> wrote: > > > > > I'm new to the Asterisk world, and I'm trying to use AstLinux to > > > replicate > > an existing voicemail environment, and I have several configuration > > questions. Is this the proper forum for these questions, or do I send > > the questions somewhere else? > > > > > > Thanks. > > > Tim. > > > > Hi Tim, > > > > First, using AstLinux as a dedicated voicemail server, using a small > > x86 appliance and SSD storage or Virtual Machine Guest is a good approach. > > > > This mailing list is mostly dedicated to AstLinux Project specific > > questions, Asterisk voicemail.conf, sip.conf and extensions.conf > > configurations are best asked in the Asterisk support groups. If you > > have things all but working and have reached a brick wall using > > AstLinux ... you can give this list a try. > > > > Keep in mind that using AstLinux, you will be required to generate the > > base extensions.conf text file for yourself, AstLinux has a basic web > > interface and "Users" tab that can help manage your voicemail users. > > As a starting point you might spin-up the "Guest VM x86-64bit (Video > > Console)" Install ISO in a virtual machine to give you a playground to > > test before purchasing any hardware. > > > > Alternatively, if coding a extensions.conf is not your cup-of-tea you > > might query this mailing list for off-line consulting help. > > > > Here is a reference to give you the flavor of the configuration ... > > > > Configuring Voice Mail Boxes > > https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxe > > s > > > > Lonnie > > ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
