I’m new to FreePBX. I managed to figure out how to do the initial recorded greeting but from here, how do I make it so that it routes to my voicemail if they press 1? After they press 1, is there a way to provide a WAV that says “please leave message after the beep”?
Also, where do I specificy SMTP settings so that I can get emails of voicemail recordings?
Thanks in advance. This is incredible software.
I would suggest you look at the series of Crosstalk Solutions videos at this link and learn about FreePBX.
Thanks. I had already viewed the applicable videos from his series. I followed his video instructions precisely. The issue I’m having now is that it doesn’t go to voicemail when I press 1. The IVR works but the Extension does not end up at voicemail. If I press any numeric input key when calling, it just tells me it’s an invalid response and eventually it hangs up. *67 or *68 has no effect to configure voicemail. This leads me to believe that the voicemail component is not running.
I tried getting into the Asterik command prompt and used “voicemail reload” but it says
“no such command 'voicemail reload”. Is the voicemail component functioning?
Applications|extension|Voicemail|VmX Locator
tick enabled, (choose unavailable or busy) and set 1 to go to voicemail of choice
Sip Sip
Hello! And thank you for your guidance. I think I might have the correct settings but after the IVR, pressing 1 still does nothing. Are these the correct option settings? Also pressing *67 or *68 does nothing.
I did not understand your question. You meant to specify from your IVR, you want to route a caller to a specific mailbox, right?
First, *97 and log in to the mailbox for the extension. Follow the prompts and record your greetings and set the password.
Applications|IVR|(at the bottom) IVR Entries
enter the digit ‘1’ and make the destination ‘Voicemail’, then assign the mailbox. The mailbox should already be set up and in that process you record the greeting.
The information I gave was for once you are in a user’s mailbox you get a ‘mini-IVR’ for your user to route calls.
SMTP settings are configured in System Admin. You should really buy a perpetual license to make it a LOT easier to configure SMTP (and other important things).
When I press *97 I get “we have not received a valid response, please try again”. I get no further prompts. It just hangs up not too long thereafter.
The intention is: IVR: “Welcome to XYZ, press 1 to leave a voicemail for us” → “beep” → blah blah blah
I took a look at all the addons here: /www.freepbx.org/add-ons/
Which of this is the SMTP function you were referring to?
Thanks for this. I verified this was the configuration I had already set up. I’m still at a loss why I can’t configure the voicemail.
Any idea why after doing the above i still cannot voicemail to work?
Maybe it cannot “hear” your button presses (1-way audio)?
Providing Great Debug - Support Services - Documentation (freepbx.org)
When I call, and if I press 1 right away, it cancels playing the initial greeting. I am assuming that it is processing some kind of input. But after I press 1, it just goes silent.
Also I ran some of the debug commands in your link above and this is what I got:
amportal a dbug
!******! 'amportal' IS DEPRECIATED. PLEASE USE 'fwconsole' !******!
FORWARDING ALL COMMANDS TO 'fwconsole', OUTPUT WILL BE MODIFIED
CONTINUING TO RELY ON AMPORTAL WILL HAVE UNDESIRED CONSEQUENCES
+-----------------------+
| FreePBX Notifications |
+-----------------------+
An attempt to send an email to "" with update notifications failed
The FreePBX project is collecting anonymous browser statistics using google analytics. These are used to focus development efforts based on real user input. All information is anonymous. You can disable this in Advanced Settings with the Browser Stats setting.
You are missing support for the following HTML5 codecs: m4a. To fully support HTML5 browser playback you will need to install programs that can not be distributed with FreePBX. If you'd like to install the binaries needed for these conversions click 'Resolve' in the lower left corner of this message. You can also safely ignore this message but browser playback might not work in your browser.
The default bind ports for FreePBX have changed. Please keep this is mind while configuring your devices. You can change this in SIP Settings. CHAN_PJSIP is: 5060, CHAN_SIP is: 5160
OUT > ==> /var/log/asterisk/freepbx_dbug <==
ERR > tail: ERR > cannot open '/var/log/httpd/error_log' for readingERR > : No such file or directoryERR >
OUT >
==> /var/log/asterisk/freepbx_security.log <==
[2023-06-30 18:04:01] [freepbx_security.NOTICE]: Authentication failure for rebel from 98.200.128.125 [] []
[2023-07-04 18:34:11] [freepbx_security.NOTICE]: Authentication failure for admin from 45.155.91.184 [] []
[2023-07-24 08:23:05] [freepbx_security.NOTICE]: Authentication failure for admin from 189.15.3.151 [] []
[2023-07-31 16:16:24] [freepbx_security.NOTICE]: Authentication failure for rebel from 98.200.128.125 [] []
[2023-07-31 16:16:28] [freepbx_security.NOTICE]: Authentication failure for jon from 98.200.128.125 [] []
[2023-07-31 16:52:33] [freepbx_security.NOTICE]: Authentication failure for rebel from 98.200.128.125 [] []
[2023-07-31 16:52:39] [freepbx_security.NOTICE]: Authentication failure for jon from 98.200.128.125 [] []
[2023-07-31 16:56:38] [freepbx_security.NOTICE]: Authentication failure for rebel from 98.200.128.125 [] []
[2023-08-10 18:59:31] [freepbx_security.NOTICE]: Authentication failure for admin from 98.200.128.125 [] []
==> /var/log/asterisk/freepbx.log <==
[2023-08-26 05:53:51] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:53:51] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:54:02] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:54:02] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:55:01] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:55:01] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:56:02] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:56:02] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:56:32] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:56:32] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
OUT > [2023-08-26 05:57:01] [freepbx.INFO]: Deprecated way to add Console commands for module backup, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2023-08-26 05:57:01] [freepbx.INFO]: Deprecated way to add Console commands for module voicemail, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
^C
tail -f /var/log/asterisk/full
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx_builtins.c: Goto (from-sip-external,s,5)
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00001c18", "TIMEOUT(absolute)=15") in new stack
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] func_timeout.c: Channel will hangup at 2023-08-26 05:57:51.100 GMT.
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:6] Set("PJSIP/anonymous-00001c18", "receveip=pjsip,remote_addr") in new stack
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:7] Log("PJSIP/anonymous-00001c18", "WARNING,"Rejecting unknown SIP connection from 45.93.16.211:50218"") in new stack
[2023-08-26 05:57:36] WARNING[476268][C-00001c19] Ext. s: "Rejecting unknown SIP connection from 45.93.16.211:50218"
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:8] Answer("PJSIP/anonymous-00001c18", "") in new stack
[2023-08-26 05:57:36] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:9] Wait("PJSIP/anonymous-00001c18", "2") in new stack
[2023-08-26 05:57:38] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:10] Playback("PJSIP/anonymous-00001c18", "ss-noservice") in new stack
[2023-08-26 05:57:38] VERBOSE[476268][C-00001c19] file.c: <PJSIP/anonymous-00001c18> Playing 'ss-noservice.ulaw' (language 'en')
[2023-08-26 05:57:43] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:11] PlayTones("PJSIP/anonymous-00001c18", "congestion") in new stack
[2023-08-26 05:57:43] VERBOSE[476268][C-00001c19] pbx.c: Executing [s@from-sip-external:12] Congestion("PJSIP/anonymous-00001c18", "5") in new stack
[2023-08-26 05:57:48] VERBOSE[476268][C-00001c19] pbx.c: Spawn extension (from-sip-external, s, 12) exited non-zero on 'PJSIP/anonymous-00001c18'
[2023-08-26 05:57:48] VERBOSE[476268][C-00001c19] pbx.c: Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00001c18", "") in new stack
[2023-08-26 05:57:48] VERBOSE[476268][C-00001c19] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on 'PJSIP/anonymous-00001c18'
[2023-08-26 05:58:01] SECURITY[1150] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2023-08-26T05:58:01.917+0000",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x7f9120005aa0",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/43672",UsingPassword="0",SessionTV="2023-08-26T05:58:01.917+0000"
[2023-08-26 05:58:14] ERROR[229350] pjproject: sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[2023-08-26 05:58:18] ERROR[20333] res_pjsip.c: Unable to create outbound OPTIONS request to endpoint twilio as URI 'sip:54.244.51.0/30:5060' is not valid
[2023-08-26 05:58:18] ERROR[20333] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:54.244.51.0/30:5060 on AOR twilio
[2023-08-26 05:59:01] SECURITY[1150] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2023-08-26T05:59:01.409+0000",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x7f911c007ec0",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/42972",UsingPassword="0",SessionTV="2023-08-26T05:59:01.409+0000"
[2023-08-26 05:59:03] VERBOSE[229350] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '209.150.30.12'
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [9009442039961628@from-sip-external:1] NoOp("PJSIP/anonymous-00001c19", "Received incoming SIP connection from unknown peer to 9009442039961628") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [9009442039961628@from-sip-external:2] Set("PJSIP/anonymous-00001c19", "DID=9009442039961628") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [9009442039961628@from-sip-external:3] Goto("PJSIP/anonymous-00001c19", "s,1") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx_builtins.c: Goto (from-sip-external,s,1)
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00001c19", "1?setlanguage:checkanon") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx_builtins.c: Goto (from-sip-external,s,2)
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00001c19", "CHANNEL(language)=en") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00001c19", "1?noanonymous") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx_builtins.c: Goto (from-sip-external,s,5)
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00001c19", "TIMEOUT(absolute)=15") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] func_timeout.c: Channel will hangup at 2023-08-26 05:59:18.204 GMT.
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:6] Set("PJSIP/anonymous-00001c19", "receveip=pjsip,remote_addr") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:7] Log("PJSIP/anonymous-00001c19", "WARNING,"Rejecting unknown SIP connection from 45.93.16.211:52500"") in new stack
[2023-08-26 05:59:03] WARNING[476342][C-00001c1a] Ext. s: "Rejecting unknown SIP connection from 45.93.16.211:52500"
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:8] Answer("PJSIP/anonymous-00001c19", "") in new stack
[2023-08-26 05:59:03] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:9] Wait("PJSIP/anonymous-00001c19", "2") in new stack
[2023-08-26 05:59:05] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:10] Playback("PJSIP/anonymous-00001c19", "ss-noservice") in new stack
[2023-08-26 05:59:05] VERBOSE[476342][C-00001c1a] file.c: <PJSIP/anonymous-00001c19> Playing 'ss-noservice.ulaw' (language 'en')
[2023-08-26 05:59:10] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:11] PlayTones("PJSIP/anonymous-00001c19", "congestion") in new stack
[2023-08-26 05:59:10] VERBOSE[476342][C-00001c1a] pbx.c: Executing [s@from-sip-external:12] Congestion("PJSIP/anonymous-00001c19", "5") in new stack
[2023-08-26 05:59:15] VERBOSE[476342][C-00001c1a] pbx.c: Spawn extension (from-sip-external, s, 12) exited non-zero on 'PJSIP/anonymous-00001c19'
[2023-08-26 05:59:15] VERBOSE[476342][C-00001c1a] pbx.c: Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00001c19", "") in new stack
[2023-08-26 05:59:15] VERBOSE[476342][C-00001c1a] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on 'PJSIP/anonymous-00001c19'
[2023-08-26 05:59:18] ERROR[229350] res_pjsip.c: Unable to create outbound OPTIONS request to endpoint twilio as URI 'sip:54.244.51.0/30:5060' is not valid
[2023-08-26 05:59:18] ERROR[229350] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:54.244.51.0/30:5060 on AOR twilio
[2023-08-26 05:59:23] SECURITY[1150] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2023-08-26T05:59:23.300+0000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="1000",SessionID="e5f4a117884790e4f7a000",LocalAddress="IPV4/UDP/192.168.128.4/5060",RemoteAddress="IPV4/UDP/96.91.177.245/5060",Challenge=""
[2023-08-26 05:59:23] NOTICE[229350] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected]>' failed for '96.91.177.245:5060' (callid: e5f4a117884790e4f7a000) - Failed to authenticate
[2023-08-26 05:59:23] SECURITY[1150] res_security_log.c: SecurityEvent="ChallengeResponseFailed",EventTV="2023-08-26T05:59:23.461+0000",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="1000",SessionID="e5f4a117884790e4f7a000",LocalAddress="IPV4/UDP/192.168.128.4/5060",RemoteAddress="IPV4/UDP/96.91.177.245/5060",Challenge="1693029563/3edc6e8ee05dc6aa04fe6c22aac10cb4",Response="4d1dc315505fc3827af6adcc7eb69411",ExpectedResponse=""
[2023-08-26 05:59:23] NOTICE[20333] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected]>' failed for '96.91.177.245:5060' (callid: e5f4a117884790e4f7a000) - Failed to authenticate
[2023-08-26 05:59:23] SECURITY[1150] res_security_log.c: SecurityEvent="ChallengeResponseFailed",EventTV="2023-08-26T05:59:23.622+0000",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="1000",SessionID="e5f4a117884790e4f7a000",LocalAddress="IPV4/UDP/192.168.128.4/5060",RemoteAddress="IPV4/UDP/96.91.177.245/5060",Challenge="1693029563/3edc6e8ee05dc6aa04fe6c22aac10cb4",Response="20ac583a5cd44d52ce8d5b130788348e",ExpectedResponse=""
Please follow the directions in the provided link and provide a trace of an impacted call via Pastebin.