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eviljafar

join:2007-04-10
Montreal, QC

[Asterisk] Setting up voicemail with FreePBX

I'm trying to set up voicemail with FreePBX 2.11, and not succeeding. I found the voicemail settings in the GUI under Applications > Extensions and made changes there. I looked but didn't find anything under Settings > Voicemail Admin that looked like I should change.

I then dialed in from the extension and recorded greetings. I am using Callcentric as my inbound trunk so I logged into the customer portal and set voicemail to answer after 60 seconds, which is the maximum I can choose, and I don't see a disable option.

When I place a test call it doesn't go to voicemail, instead I hear a 3 tone response a couple of times.

I googled for a while but didn't come up with an answer. All useful sounding posts suggested I edit config files, but when I open those files FreePBX has comments at the start telling me not to edit them, and that I should change settings through the GUI.

Can someone point me in the direction of something like a dummy's guide to FreePBX voicemail?

Thanks.


SCADAGeo

join:2012-11-08
N California
kudos:2

said by eviljafar:

set voicemail to answer after 60 seconds, which is the maximum I can choose, and I don't see a disable option.

Voice mail can be removed from the CallCentric account, but I like to keep it active because it can serve as an indicator of my PBX's health.

CallCentric voice mail is configured to e-mailed the VM, so if I receive an e-mailed CallCentric voice mail, I know there is an issue with my PBX.

 
said by eviljafar:

When I place a test call it doesn't go to voicemail, instead I hear a 3 tone response a couple of times.

You didn't mention which distribution of Asterisk/FreePBX that you're using, but you may be hearing the ss-noservice Special Information Tone (SIT) played by the PBX.

This is a defensive measure in case a call comes in via an unknown or unconfigured route.

CallCentric's calls may come from a number of IPs, so I found this post regarding multiple CallCentric DIDs to be helpful when I was troubleshooting my initial setup.

 
said by eviljafar:

All useful sounding posts suggested I edit config files, but when I open those files FreePBX has comments at the start telling me not to edit them, and that I should change settings through the GUI.

My preference is to copy the code (context/function) that I want to modify into "extensions_override_freepbx.conf".

This has two main purposes: future proofing (updating to latest version of freepbx doesn't wipe out your code), and easy reversion/debugging.

 
said by eviljafar:

Can someone point me in the direction of something like a dummy's guide to FreePBX voicemail?

You may find this useful: O'Reilly OPFS: Asterisk: The Definitive Guide, 4th Edition.


Trimline
Premium
join:2004-10-24
Windermere, FL
reply to eviljafar

Along with the above, make sure that your inbound route is your 1777XXXXXXX number and not your DID.


eviljafar

join:2007-04-10
Montreal, QC
reply to SCADAGeo

Thanks for your reply, in the end it was a code 18 error :$

I don't think it's going to work out for me though as I often use voice mail to email and I have since found out that my ISP blocks outbound port 25 and I don't have access to a mail server elsewhere.



Trimline
Premium
join:2004-10-24
Windermere, FL
Reviews:
·ObiVoice
·Bright House
·Callcentric
·voip.ms

Look through this article : »pbxinaflash.com/community/index.···ail.974/

You can use an email relay


SCADAGeo

join:2012-11-08
N California
kudos:2
reply to eviljafar

said by eviljafar:

Thanks for your reply, in the end it was a code 18 error :$

Result code 18 is AST_CAUSE_NO_USER_RESPONSE (no answer / time out), and the dial plan (context/function) determines the SIT that is played.

 
said by eviljafar:

I don't think it's going to work out for me though as I often use voice mail to email and I have since found out that my ISP blocks outbound port 25 and I don't have access to a mail server elsewhere.

said by Trimline:

Look through this article : »pbxinaflash.com/community/index.···ail.974/

You can use an email relay

It's a common obstacle, and there is a lot of good information in the link Trimline See Profile provided.

eviljafar

join:2007-04-10
Montreal, QC

said by SCADAGeo:

Result code 18 is AST_CAUSE_NO_USER_RESPONSE (no answer / time out), and the dial plan (context/function) determines the SIT that is played.

Oh sorry to make you look that up, I was joking, code 18 is a slang term meaning the problem is 18 inches from the screen, ie the problem was me.

eviljafar

join:2007-04-10
Montreal, QC
reply to Trimline

Thanks, I'll check it out.


SCADAGeo

join:2012-11-08
N California
kudos:2
reply to eviljafar

said by eviljafar:

I was joking