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Two extensions with same mailbox

You can “link” two mailboxes which reside on the same server by creating a symlink between them.

For example, create two extensions each with mailboxes (extension 1000 and 2000). Dial *1000, wait for the voicemail greeting to start and then hangup. Do the same thing for extension 2000 (dial *2000). This will initialize the two mailboxes. Then from the shell, navigate to:

/var/asterisk/spool/voicemail/default

You will see a folder for named “1000” and one named “2000”. Delete one of them:

rm -rf 2000

Create a symlink from 2000 to 1000:

ln -s 1000 2000

That’s it! Any voicemail left in either mailbox will be available through either mailbox. MWI will also be reflected on both extensions.

https://www.voip-info.org/forum/threads/two-extensions-one-vm-box.3390/
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MariaDB add new user

[user@hostname] # mysql
MariaDB [(none)]>CREATE USER 'user1'@localhost IDENTIFIED BY 'password1';
MariaDB [(none)]>SELECT User FROM mysql.user; 
MariaDB [(none)]>GRANT ALL PRIVILEGES ON *.* TO 'user1'@localhost IDENTIFIED BY 'password1'; 
MariaDB [(none)]>GRANT ALL PRIVILEGES ON 'yourDB'.* TO 'user1'@localhost;  
MariaDB [(none)]>FLUSH PRIVILEGES;  
MariaDB [(none)]>SHOW GRANTS FOR 'user1'@localhost;

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Linksys/Cisco SPA3102 with Freepbx & BT

This guide works great, with the one amendment; PJSIP Registration should be set to “none”.

https://frag.co.uk/2017/08/14/cisco-spa-3102-freepbx-uk-caller-id/

Critical config info is as follows:

SPA-3102 Configuration

Admin login > Advanced > Voice > PSTN Line:

  • Line Enable: Yes
  • SIP Transport: UDP
  • SIP Port: 5060
  • SIP Remote-Party-ID: Yes
  • Auth Invite: Yes
  • Proxy: [set to FreePBX IP]
  • Register: No
  • User ID: pstn_fxo
  • Auth ID: pstn_fxo
  • Use Auth ID: Yes
  • Password: password
  • Line 1 VoIP Caller DP: None
  • PSTN Caller Default DP: 1
  • Dial Plan 1: (S0<:pstn_fxo@192.168.0.181>)
    Including brackets – and where 192.168.0.181 is your FreePBX IP

FreePBX Configuration

Create a new PJSIP Trunk

  • Trunk Name: <My DID>
  • Outbound Caller ID: <My DID>
  • Maximum Channels: 1
  • Username:  <My DID>
  • Password: password
  • Authentication: Outbound
  • Registration: Send
  • SIP Server: [set to IP address of SPA-3102]
  • SIP Server Port: 5060
  • Context: from-pstn
  • Contact User:  <My DID>  (thanks to Aly)

Inbound Route

  • Set up a default route with Destination as a Ring Group

Outbound Route

  • Route Name: POTS Outgoing
  • Route CID: <My DID>
  • Trunk Sequence for Matched Routes:  <My DID>

More notes:

https://basichelp.sipgate.co.uk/hc/en-gb/articles/206289069-UK-Regional-Settings-Cisco-Linksys-Sipura-Adaptors-