So just wanted to thank Lorne and Daniel for their help on this. For the MWI and Voicemail redirection both are working. Here are the steps that are required to get them going:
There are two components the first as mentioned above is to add the “unsolicited_mailbox” line to the “PEER Details” for your Trunk.
Go to "Connectivity > Trunks"
Under “PEER Details” add the last line in the config below
host=atlanta.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)
secret=johnspassword ;your password
username=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;(Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
unsolicited_mailbox=65000 ; 65000 being your mailbox ID which is different from your SIP account ID. Login to you voip.ms account and look under "DID Numbers > Voicemail"
Next you need to go to “Applications > Extensions” and you have to go to “Mailbox” and then add the following line:
Again replace 65000 with your mailbox ID.
Now I am working with Avaya phones as well which make this a little more iffy, they don’t seem to love a non-Avaya environment. I will post some additional details below on how to get them registering with Freepbx.
Now to get your voicemail calling for *97 redirected to VOIP.ms you will have to either go to “Admin > Feature Codes” go down to the last one voicemail and change it to something else I used *88 and save and apply config and then disable it.
Next go back to “Connectivity > Trunks” and create a new Outbound rule and name it whatever you want. In the dialplan section just add *97 to the first field after prefix. I also added *xx as a secondary one but I don’t think it’s necessary. Ok save and apply config.
Lastly in the top Right hand corner where you see your outbound routes, drag the new route to be first in order and again apply config.
Some Notes on Avaya Phones and asterisk/freepbx
First you need to update your phones to a SIP firmware you can download them from Avaya’s website.
just an example:
Setup an HTTP server and make sure that port 80 is open and available. I used HFS
In your Avaya phone you need to enter the config settings. In order to do this you can either hit the * key when you see the program button when the phone is booting up. Or if it is already booted up you can use the following key sequence:
That will get you into the config menu and you can then you can go to ADDR and setup the address for the HTTP server.
You will also need to setup a 46xxsettings.txt file where all your settings are stored for the phone:
This is an example of some possible settings to use:
post by salabie has the necessary details:
SET DOMAIN 192.168.1.100
SET SIPDOMAIN 192.168.1.100
SET SIPPORT 5060
SET SIP_CONTROLLER_LIST 192.168.1.100:5060;transport=tcp
SET SIPREGPROXYPOLICY alternate
SET CONFIG_SERVER_SECURE_MODE 0
SET SIPPROXYSRVR 192.168.1.100
SET SIPSIGNAL 1
SET SIP_PORT_SECURE 5061
SET ENABLE_AVAYA_ENVIRONMENT 0
these are the most important.
Next in your extension settings in Freebpx you have to turn off “NAT” so “Nat Mode” set to “No- (no)” and also set your Protocol “Transport” to “TCP Only”.
Also go to “Settings > Asterisk SIP Settings” then in the top right hand corner select “Chan SIP” and then locate the “Other SIP Settings” field and add the following:
tcpenable = yes
Any questions shoot me a PM. Thanks again to all of those who helped, I appreciate it. I hope some of the info here is useful to anyone else.