Voicemail activation

Says extension 2000 that the voicemail is not activated, but under extensions it says its turned on. how do you now activate it?

Dial *97 from the extension or *982000 from anywhere else.

tried that and it says that number is invalid.

actually says the number you have dial is not in service. Freepbx shows it as inactive.

Do *97 and *98 work from other extensions?

everything in your dialplan can be tested at the horses mouth with

dialplan show [email protected]

so (only use ‘dialplan show @’ if you want everything)

dialplan show *[email protected]

will show what happens when you dial *97 from an internal extension

only just starting to set this up, but the system says its inactive

is this done from the front end and not the GUI?

I get

dialplan show *[email protected]
[ Included context 'app-vmmain' created by 'pbx_config' ]
  '*97' =>          1. Macro(user-callerid,)                      [extensions_additional.conf:2935]                                                                                           
                    2. Set(CONNECTEDLINE(name-charset,i)=utf8)    [extensions_additional.conf:2936]                                                                                           
                    3. Set(CONNECTEDLINE(name,i)=My Voicemail)    [extensions_additional.conf:2937]                                                                                           
                    4. Set(CONNECTEDLINE(num,i)=${AMPUSER})       [extensions_additional.conf:2938]                                                                                           
                    5. Answer()                                   [extensions_additional.conf:2939]                                                                                           
                    6. Wait(1)                                    [extensions_additional.conf:2940]                                                                                           
                    7. Macro(get-vmcontext,${AMPUSER})            [extensions_additional.conf:2941]                                                                                           
     [check]        8. Set(VMBOXEXISTSSTATUS=${IF(${VM_INFO(${AMPUSER}@${VMCONTEXT},exists)}?SUCCESS:FAILED)}) [extensions_additional.conf:2942]                                              
                    9. GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?mbexist) [extensions_additional.conf:2943]                                                                                
                    10. VoiceMailMain()                           [extensions_additional.conf:2944]                                                                                           
                    11. GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret) [extensions_additional.conf:2945]                                                               
                    12. Macro(hangupcall,)                        [extensions_additional.conf:2946]                                                                                           
     [mbexist]      109. GotoIf($["${DB(AMPUSER/${AMPUSER}/novmpw)}"!=""]?novmpw:vmpw) [extensions_additional.conf:2947]                                                                      
     [novmpw]       110. Noop(Verifying channel ${CHANNEL} is actually ${AMPUSER}) [extensions_additional.conf:2948]                                                                          
                    111. Set(DEVICES=${DB(AMPUSER/${AMPUSER}/device)}) [extensions_additional.conf:2949]                                                                                      
                    112. ExecIf($["${DEVICES}" = ""]?Set(DEVICES=${AMPUSER})) [extensions_additional.conf:2950]                                                                               
                    113. ExecIf($["${DEVICES:0:1}" = "&"]?Set(DEVICES=${DEVICES:1})) [extensions_additional.conf:2951]                                                                        
                    114. While($["${SET(DEV=${SHIFT(DEVICES,&)})}" != ""]) [extensions_additional.conf:2952]                                                                                  
                    115. GotoIf($["${DB(DEVICE/${DEV}/dial)}" = "${CUT(CHANNEL,-,1)}"]?vmpwskip) [extensions_additional.conf:2953]                                                            
                    116. EndWhile()                               [extensions_additional.conf:2954]                                                                                           
                    117. Noop(Channel ${CHANNEL} is NOT ${AMPUSER} forcing VM Password) [extensions_additional.conf:2955]                                                                     
     [vmpw]         118. VoiceMailMain(${AMPUSER}@${VMCONTEXT})   [extensions_additional.conf:2956]                                                                                           
                    119. Goto(vmend)                              [extensions_additional.conf:2957]                                                                                           
     [vmpwskip]     120. VoiceMailMain(${AMPUSER}@${VMCONTEXT},s) [extensions_additional.conf:2958]                                                                                           
     [vmend]        121. GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret) [extensions_additional.conf:2959]                                                              
                    122. Macro(hangupcall,)                       [extensions_additional.conf:2960]                                                                                           
     [playret]      123. Playback(beep&you-will-be-transfered-menu&silence/1) [extensions_additional.conf:2961]                                                                               
                    124. Goto(${IVR_CONTEXT},return,1)            [extensions_additional.conf:2962]                  

Not sure what you mean by “front end” , but yes you can do this from any “shell” including ssh, local terminal or even (god forbid) putty. (If windoze bound, look into WSL (windows subsystem linux or somesuch double speak) )

what will i be looking for then?

No matter what i do i get The number you have dialed is not in service, this is any outbound call or *97 for voicemail. what am i missing here?

Doesn’t appear that my server is making it to the outside world. how do i verify it reaching the outside world? doesn’t appear to be registering with Vitelity.

dialplan show [email protected]

dialplan show [email protected]

dialplan show [email protected]

dialplan show *[email protected]

dialplan show *[email protected]

each will parse what you have dialed and from whatever context (here from-internal)

If there is no good dialplan , it will show you.

why for what ever reason does everything give me the The number you have dialed is not in service. also how do i verify that the server is connecting to vitelity?

vitelity registration shows the sub account as offline and not connected. im thinking this is a problem.

i get command not found when typing those commands in to the prompt. i went to the server and signed in with the ROOT user name and password and then type that command. is that where im supossed to do it?

that is for the asterisk cli, for a shell (bash, putty ssh),

rasterisk -x ‘dialplan show xxxx’

you need to fulfill any needed dialplan in your preferred context or it will fail

You seem to be asking about several different topics. Please tell us a little about your system.
Distro? If not, how built?
Cloud? If so, whose? If not, physical or virtual? If virtual, platform?
What is your extension device (some clients may interpret star codes locally or block numbers that don’t match their dialplan)?
Find the simplest thing that fails:
*65 (announces your extension number).
*43 (echo test).
Calling from one extension to another.
Calling 18004377950.

At the Asterisk command prompt, type
pjsip set logger on
if using pjsip extensions or trunks, and/or
sip set debug on
if using chan_sip extensions or trunks.
Make the simplest failing test call. Go to Reports -> Asterisk Logfiles and find the section relevant to your failing call. Paste it at https://pastebin.freepbx.org and post the link here. Also, report what you heard and what your phone displayed.

Well here is where i am, I am completely new to this Freepbx and setting up my own PBX. So i downloaded the Freepbx file to a thumb drive and i loaded it to a small server i had. everything went well. I was watching some videos that stepped my thought it. So that said. i also have Vitelity. I have followed all steps to get it online and im not successful. the * codes all come back as the number you have dialed is not in service please check the number and try again. I am using sangoma s500 phones. i noticed on the phone its self the little phone icon has a red line thought it. I also noticed when ever i dial a number i need to press send. I also noticed in the above report that i pulled that the system is not registering with Vitelity.

Log from the asterisk logfile report i pulled:

[2020-07-11 15:45:41] ERROR[12625] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2020-07-11 15:45:41] NOTICE[12625] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #384)

Were the phones configured with Endpoint Manager, or manually? pjsip extensions or chan_sip? In Reports -> Asterisk Info, what status is shown for your extensions?

From a shell prompt, what does
ping inbound6.vitelity.net

Make a test call to *65, paste the complete log for it at pastebin.freepbx.org and post the link here.