Fusionpbx Docs

FusionPBX Documentation
Mark J Crane
Len Graham
Apr 03, 2020

Contents
1 FusionPBX 3
1.1 Benefits of FusionPBX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.2 FusionPBX Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.3 Our Ecosystem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2 Getting Started 5
2.1 Getting Started . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1.1 Training . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1.1.1 Free . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1.1.2 Paid Training and technical support . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1.2 Quick Install . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1.3 Let’s Encrypt . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.1.3.1 Dehydrated (Recommended) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.1.4 Provision . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.1.4.1 Automatic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.1.4.2 Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.1.4.3 Advanced > Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.1.4.4 Phone Screen Capture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
2.1.4.5 Phone Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
2.1.5 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.1 FusionPBX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.2 Firewall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.3 SSL / TLS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.4 Upgrade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.5 XML RPC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.1.5.6 Fail2ban . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
2.1.5.7 SSH . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
2.1.6 Backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
2.1.6.1 Command Line . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
2.1.6.2 Crontab (optional) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
2.1.7 Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
2.1.8 Firewall . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.1.8.1 Iptables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.1.8.2 Fail2ban . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
2.1.8.3 PF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
2.1.9 Languages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
i
2.1.9.1 Fusionpbx Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
2.1.9.2 FreeSWITCH Sound Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
2.1.9.3 app_languages.php . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
3 Home 31
3.1 Home . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
3.1.1 Account Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
3.1.2 Dashboard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
3.1.2.1 Dashboard Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
3.1.3 Logout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
4 Accounts 35
4.1 Accounts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
4.1.1 Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
4.1.1.1 Device Vendors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
4.1.1.2 Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
4.1.2 Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
4.1.2.1 Basic Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
4.1.2.2 Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
4.1.3 Gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
4.1.3.1 Basic Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
4.1.3.2 Advanced Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
4.1.4 Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
4.1.4.1 Users Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
5 Dialplans 45
5.1 Dialplans . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
5.1.1 Destinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
5.1.1.1 Destinations Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
5.1.2 Dialplan Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
5.1.3 Dialplan Details . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
5.1.3.1 Global . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
5.1.3.2 Domain Specific . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
5.1.4 Dialplan Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
5.1.5 Inbound Routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
5.1.5.1 Edit/Add Inbound Routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
5.1.5.2 XML example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
5.1.6 Outbound Routes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
5.1.6.1 Pin Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
5.1.7 Advanced Dialplans . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
5.1.7.1 Adding a Dialplan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
5.1.7.2 Edit a Dialplan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
5.1.7.3 Enable a Dialplan Destination . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
5.1.7.4 Dialplan example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
6 Applications 83
6.1 Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
6.1.1 Bridges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
6.1.1.1 Bridge Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
6.1.2 Call block . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
6.1.3 Call Broadcast . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
6.1.4 Call Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
6.1.4.1 Call Center Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
6.1.4.2 Call Center Agents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
6.1.4.3 Call Center Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
ii
6.1.4.4 Agents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.5 Music On Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.6 Record . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.7 Time base score . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.8 Max Wait Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.9 Max Wait Time with No Agent . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.10 Max Wait Time with No Agent Time Reached . . . . . . . . . . . . . . . . . . . . 89
6.1.4.11 Timeout Action . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
6.1.4.12 Tier Rules Apply . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.13 Tier Rule Wait Second . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.14 Tier Rule Wait Multiply Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.15 Tier Rule No Agent No Wait . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.16 Discard Abandoned After . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.17 Abandoned Resume Allowed . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.18 Caller ID Name Prefix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.19 Announce Sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.20 Announce Frequency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.21 Exit Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
6.1.4.22 Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
6.1.4.23 Agent Call Center Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
6.1.5 Call Detail Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
6.1.5.1 Post Dial Delay (PDD) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
6.1.5.2 Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
6.1.5.3 Possible issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
6.1.5.4 XML CDR configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
6.1.5.5 Harddrive space usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
6.1.5.6 CDR Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
6.1.6 Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
6.1.6.1 Call Flow Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
6.1.7 Call Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
6.1.8 Call Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
6.1.8.1 Call Forward and Do No Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
6.1.9 Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97
6.1.9.1 Conference Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
6.1.9.2 Enable Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
6.1.10 Conference Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
6.1.10.1 Conference Center Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
6.1.10.2 Conference Center Rooms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
6.1.11 Conference Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
6.1.11.1 Default Conference Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
6.1.12 Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
6.1.12.1 Default Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
6.1.13 Contacts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
6.1.14 Fax Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
6.1.14.1 Fax Server Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
6.1.14.2 New . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
6.1.14.3 Inbox . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
6.1.14.4 Sent . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
6.1.14.5 Fax ATA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
6.1.14.6 FAX Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
6.1.15 Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
6.1.15.1 Follow Me Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
6.1.16 GS Wave . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
6.1.17 IVR Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
iii
6.1.17.1 IVR Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
6.1.18 Operator Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
6.1.18.1 Operator Panel Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
6.1.19 Phrases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
6.1.20 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
6.1.20.1 Music on Hold Tips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
6.1.21 Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
6.1.22 Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
6.1.22.1 Create a Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
6.1.22.2 Edit Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
6.1.22.3 Applying Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
6.1.22.4 Recordings Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
6.1.23 Ring Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
6.1.23.1 Ring Group Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
6.1.24 Streams . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
6.1.25 Time Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
6.1.25.1 Time Conditions Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
6.1.25.2 Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
6.1.26 Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
6.1.26.1 Voicemail Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
6.1.26.2 Voicemail Transcription . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134
7 Status 137
7.1 Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
7.1.1 Active Call Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
7.1.2 Active Calls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
7.1.3 Active Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
7.1.4 Active Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
7.1.5 Agent Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
7.1.6 CDR Statistics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
7.1.6.1 Definitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
7.1.7 Emails . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
7.1.8 Extension Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
7.1.8.1 Definitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
7.1.9 Log Viewer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
7.1.10 Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
7.1.11 Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
7.1.12 SIP Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
7.1.13 System Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
7.1.14 Traffic Graph . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
8 Advanced 149
8.1 Advanced . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
8.1.1 Access Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
8.1.1.1 Access Control Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
8.1.2 App Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
8.1.3 Command . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
8.1.4 Databases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
8.1.5 Default Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
8.1.5.1 Adminer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
8.1.5.2 Cache . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
8.1.5.3 Call Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
8.1.5.4 CDR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
8.1.5.5 Dashboard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
iv
8.1.5.6 Destinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
8.1.5.7 Domains . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
8.1.5.8 Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
8.1.5.9 Email . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
8.1.5.10 Fax . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
8.1.5.11 Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
8.1.5.12 Ivr Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
8.1.5.13 Limit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
8.1.5.14 Login . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
8.1.5.15 Provision . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
8.1.5.16 Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
8.1.5.17 Ring Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
8.1.5.18 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
8.1.5.19 Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
8.1.5.20 Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
8.1.5.21 Theme . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165
8.1.5.22 Time Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
8.1.5.23 User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
8.1.5.24 Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
8.1.6 Domains . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
8.1.6.1 Adding a domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
8.1.6.2 Domain Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
8.1.7 Grammar Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
8.1.8 Group Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
8.1.8.1 User Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
8.1.9 Menu Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 176
8.1.10 Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 176
8.1.10.1 Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.2 Auto . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.3 Dialplan Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.4 Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.5 Event Handlers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.6 File Format Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.7 Languages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.8 Loggers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.9 Say . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.10 Speech Recognition/Text to Speech . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.11 Streams/Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.10.12 XML Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.11 Number Translations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.1.12 PHP Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178
8.1.13 Provision Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
8.1.14 Sip Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
8.1.14.1 Internal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
8.1.14.2 External . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
8.1.14.3 Internal ipv6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
8.1.14.4 External ipv6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
8.1.15 Scripts Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
8.1.16 Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
8.1.17 Transactions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
8.1.18 Upgrade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
8.1.18.1 How to Upgrade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
8.1.18.2 Move to a different Branch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
8.1.19 Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
v
8.1.19.1 Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.2 Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.3 IP Address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.4 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.5 Ringtones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.6 Sip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.7 Sip Profile:External . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.8 Sip Profile:Internal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.9 Sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.10 Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.19.11 Xmpp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
8.1.20 XML Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
9 Hardware 193
9.1 Hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
9.1.1 Auto Provision Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
9.1.1.1 Yealink . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
9.1.1.2 Polycom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
9.1.1.3 Cisco SPA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 199
9.1.1.4 Fanvil . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
9.1.1.5 Grandstream . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
9.1.1.6 Htek . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
9.1.1.7 Zoiper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
9.1.2 Manually Provision Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
9.1.2.1 Yealink . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
9.1.2.2 Polycom . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
9.1.2.3 Cisco . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 212
9.1.2.4 Fanvil . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
9.1.2.5 GrandStream . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 215
9.1.2.6 Htek . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
9.1.2.7 Zoiper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
9.1.2.8 SNOM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
9.1.2.9 Phone Screen Capture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
9.1.2.10 Product Models . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
9.1.3 Firewall Devices . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
9.1.3.1 ASUS RT-AC66U . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
9.1.3.2 Ubiquiti Edgerouter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
9.1.3.3 pfSense . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
9.1.3.4 SonicWall TZ-SOHO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 241
9.1.3.5 ZyXel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247
9.1.3.6 Cisco EA6500 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 251
10 Software 253
10.1 Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
10.1.1 Software Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
10.1.1.1 Packet Capture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
10.1.1.2 Call Quality and Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
10.1.2 Using SNGREP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
10.1.2.1 Main Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
10.1.2.2 SPAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
10.1.2.3 Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
10.1.2.4 Registration Expanded . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
10.1.2.5 Call Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
10.1.2.6 F3 Search . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
vi
10.1.2.7 F7 Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
10.1.2.8 F8 Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 259
10.1.3 TFTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
11 Additional Information 265

11.1 Additional Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
11.1.1 Voip Quality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
11.1.1.1 Voip Quality Testing Websites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
11.1.1.2 Packet Loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
11.1.1.3 Jitter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
11.1.1.4 Latency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
11.1.1.5 Wrong Configurations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
11.1.1.6 Network Attacks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
11.1.2 Freeswitch install . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
11.1.2.1 Monit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
11.1.3 SSL/TLS Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274
11.1.3.1 Configure TLS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 275
11.1.4 Testimonials . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 276
11.1.5 Password Reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 277
11.1.5.1 Old Password Reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 280
11.1.5.2 Password Hash . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
11.1.5.3 SQLite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
11.1.5.4 PostgreSQL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
11.1.5.5 Change the Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
11.1.6 Feature Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
11.1.6.1 Basic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
11.1.6.2 Call Parking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
11.1.6.3 Advanced . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.6.4 Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.6.5 Miscellaneous . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.7 Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.7.1 Announcements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.7.2 Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.7.3 Call Barge / Eavesdrop / Intercept . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
11.1.7.4 Call Block . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.5 Call Broadcast . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.6 Call Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.7 Call Detail Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.8 Call Flows (Day Night Mode) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.9 Call Forward . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.10 Call Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.11 Call Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.12 Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.13 Call Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.14 Call Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
11.1.7.15 Call Announced Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.16 Call Blind Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.17 Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.18 Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.19 Caller ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.20 Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.21 Conference Center . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.22 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.23 Contacts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
vii
11.1.7.24 Command . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
11.1.7.25 Dialplan Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.26 Dial by Name (*411) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.27 Direct Inward System Access (DISA) . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.28 Device Provisioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.29 Do Not Disturb (DND) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.30 Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.31 Extension Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.32 Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.33 Fax Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.34 Follow Me . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
11.1.7.35 Gateways . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.36 Hot Desking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.37 Inbound and Outbound Call Routing . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.38 IVR Menus (Auto Attendant) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.39 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.40 Multi-Tenant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.41 Operator Panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.42 Paging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.43 Parking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.44 Phrases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
11.1.7.45 Provider Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.46 Re-branding and Customize . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.47 Recordings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.48 Ring Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.49 Scalable and Redundant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.50 Time Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.51 User and Group Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.52 Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.53 Voicemail to Email . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.54 Voicemail Transcription . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.55 WebRTC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
11.1.7.56 Additional Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
11.1.8 Toll Allow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
11.1.8.1 PERMIT TOLL CALLS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
11.1.8.2 PREVENT TOLL CALLS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
11.1.9 Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
11.1.9.1 Default config . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
11.1.9.2 SIP ALG . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
11.1.9.3 Static IP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
11.1.9.4 UPnP or PMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
11.1.9.5 Symptoms of misconfigured NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
11.1.10 Version Upgrade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
11.1.10.1 4.4 to Master (what will become 4.6) . . . . . . . . . . . . . . . . . . . . . . . . . 291
11.1.10.2 Version 4.2 to 4.4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 292
11.1.10.3 Version 4.0 to 4.2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 293
11.1.10.4 Version 3.8 to 4.0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 295
11.1.10.5 Version 3.6 to 3.8 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 295
11.1.10.6 Version 3.5 to 3.6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
11.1.10.7 Version 3.4 to 3.5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
11.1.10.8 Version 3.3 to 3.4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
11.1.10.9 Version 3.2 to 3.3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300
11.1.10.10Version 3.1.4 to 3.2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300
11.1.10.11Version 2 to 3.0 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
viii
11.1.10.12SQLite . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
11.1.10.13ODBC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
11.1.10.14Postgres . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
11.1.10.15Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
11.1.10.16Configure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
11.1.10.17switch.conf.xml . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
11.1.11 Releases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
11.1.12 Regular Expressions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
11.1.13 PostgreSQL . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
11.1.13.1 Backup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
11.1.13.2 Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
11.1.13.3 Console . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
11.1.13.4 Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
11.1.14 Shared Line Appearance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
11.1.14.1 FusionPBX Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
11.1.14.2 Yealink SLA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
11.1.14.3 Polycom SLA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
11.1.15 CDR Archive Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 306
11.1.15.1 Move the Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
12 Contributing 311
12.1 Contributing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
13 Documentation Guide 313

13.1 Documentation Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
13.1.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
13.1.2 Getting Started . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
13.1.2.1 Getting Git Right . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
13.1.2.2 Setting up the Docs Locally . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
13.1.3 Text Formatting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
13.1.3.1 Inline markup and special characters (e.g., bold, italic, verbatim) . . . . . . . . . . 315
13.1.3.2 Headings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 316
13.1.3.3 Internal and External Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 317
13.1.3.4 List and bullets . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
13.1.4 What are directives . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
13.1.5 Code and Literal blocks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
13.1.5.1 How to include simple code . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
13.1.5.2 code-block directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
13.1.5.3 Include code with the literalinclude directive . . . . . . . . . . . . . . . . . . . . . 319
13.1.6 Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
13.1.6.1 Simple tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
13.1.6.2 Multicells tables, first method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
13.1.6.3 Multicells table, second method . . . . . . . . . . . . . . . . . . . . . . . . . . . . 321
13.1.6.4 The tabularcolumns directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 321
13.1.6.5 The csv-table directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
13.1.7 The toctree directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
13.1.8 Images and figures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
13.1.8.1 Include Images . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
13.1.8.2 Include a Figure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
13.1.9 Boxes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 324
13.1.9.1 Colored boxes: note, seealso, todo and warnings . . . . . . . . . . . . . . . . . . . 324
13.1.9.2 Topic directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 324
13.1.9.3 Sidebar directive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
13.1.10 Others . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
ix
13.1.10.1 Comments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
13.1.10.2 Substitutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 325
13.1.10.3 glossary, centered, index, download and field list . . . . . . . . . . . . . . . . . . . 326
13.1.10.4 Footnote . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
13.1.10.5 Citations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
13.1.10.6 More about aliases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 327
13.1.10.7 Intersphinx . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 328
13.1.10.8 file-wide metadata . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 328
13.1.10.9 metainformation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 328
13.1.10.10contents directives . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 328
13.1.11 Useful extensions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 329
13.1.11.1 pngmath: Maths and Equations with LaTeX . . . . . . . . . . . . . . . . . . . . . 329
13.1.11.2 TODO extension . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 330
14 Other 331
14.1 Other . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 331
14.1.1 XMPP Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 331
14.1.1.1 Enable XMPP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
14.1.2 WebRTC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 334
14.1.2.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 334
14.1.2.2 Install Steps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
14.1.3 Parking . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
Index 337
x
Contents 1
2 Contents
CHAPTER 1
FusionPBX
An open source project that provides a customizable and flexible web interface to the very powerful and highly scalable
multi-platform voice switch called FreeSWITCH.
FusionPBX will run on a variety of operating systems (Optimized for Debian 8+) and hardware of your choice.
FusionPBX provides a GUI for QR Code soft phone provisioning, unlimited extensions, voicemail-to-email, music
on hold, call parking, analog lines or high density T1/E1 circuits, and many other features. FusionPBX provides
the functionality that business need and provides corporate level phone system features to small, medium and large
businesses. Click here for the FusionPBX youtube channel .
1.1 Benefits of FusionPBX
1. Adding extra functionality to the incredibly robust FreeSWITCH VoIP Platform.

2. Makes FreeSWITCH easy to administer while at the same time still allowing you to work directly within
FreeSWITCH Command Line Interface (fs_cli) when you need to.
3. Gives your users and tenants an attractive GUI interface to interact with.
1.2 FusionPBX Features
Call Block Call Broadcast Call Flows Call Center

Call Detail Records Conference Contacts Fax Server
Follow Me Hot Desking IVR Menus Ring Groups
Multi-Tenant Music on Hold Queues Recordings
Device Provisioning Streams QR Code Provisioning SMS/MMS ready
Time Conditions WebRTC ready Voicemail and lots more. . .
3
1.3 Our Ecosystem
We are a global community that has an open and very friendly ecosystem. We encourage community engagement,
contribution and feedback. Please join us by getting involved with giving feedback, new feature ideas, helping out
with code or Documentation.
Most of the core folks who develop and use FusionPBX can be found hanging out in Freenode IRC in the #fusionpbx
channel. Come join us and meet the team.
4 Chapter 1. FusionPBX
CHAPTER 2
Getting Started
2.1 Getting Started
Welcome! Let’s install FusionPBX. Follow the menu to the left and you will have a working PBX in no time. For PDF
and Epub formats of this documentation click the bottom left on v:latest and a menu will pop-up to choose from.
Note: There are many ways to install FusionPBX depending on how you want to build your solution. What is
presented here represents the quickest, easiest, best supported way to a FusionPBX system. For advanced topics like
Bi Directional Replication or High Availability, consider attending the in person or online training at https://www.
fusionpbx.com/training.php. Additional FusionPBX training is available via Continuing Education. This is a monthly
affordable option to keep you current and ahead of the competition!
2.1.1 Training
FusionPBX offers different levels of training. We love helping people and companies succeed.
• Small family business.
• Someone that loves to tinker with technology.
• Major companies all around the world.
2.1.1.1 Free
FusionPBX has content available in this document and videos on YouTube to help get you started.
• Free Self help is available for you right here at docs.fusionpbx.com
• Videos at https://youtube.com/fusionpbx
5
2.1.1.2 Paid Training and technical support
FusionPBX offers an Admin class, Advanced class and an affordable Continuing Education class. If you want to
accelerate your understanding of how to use FusionPBX and unleash your full potential these classes are for you.
• Admin Class
– Online or Webex
– Access to Class Documentation
– Recorded video of the class for you to keep
– Interactive question and answer with the FusionPBX founder and trainer
– Covers the basics and some advanced topics with FusionPBX
– Plus much more. . .
• Advanced Class
– Online or Webex
– Access to Class Documentation
– Recorded video of the class for you to keep
– Interactive question and answer with the FusionPBX founder and trainer
– Covers advanced topics and methods
• Continuing Education
– Monthly Webex meeting on changes with FusionPBX
– Option to take the advanced or admin over again (included in the monthly price)
– Report bugs or ask questions that might not have came up until now
– Get a closer more indepth look into FusionPBX
– Know how to use new features when they are added
– Know how to upgrade to the next release version
Come join us! We look forward to meeting you!
2.1.2 Quick Install
6 Chapter 2. Getting Started

Welcome to the FusionPBX installation guide.
FusionPBX can be installed on several different operating systems. However this guide assumes you are starting with
a minimal install of Debian 9 with SSH enabled. This install has been designed to be fast, simple and modular, and
generally takes 5 minutes or less. Installation times depend on factors like CPU, RAM, disk I/O and bandwidth. Install
Video https://youtu.be/YmIht8hEHYU
1. Run the following commands as root. The script installs FusionPBX, FreeSWITCH release package and its depen-
dencies, iptables, Fail2ban, NGINX, PHP-FPM and PostgreSQL.
Start with a minimal install of Debian 9 with SSH enabled. Paste the following commands in the console window one
line at a time.
wget -O - https://raw.githubusercontent.com/fusionpbx/fusionpbx-install.sh/master/
˓→debian/pre-install.sh | sh;
cd /usr/src/fusionpbx-install.sh/debian && ./install.sh
If using Debian on Proxmox LXC containers please run the following BEFORE starting the FusionPBX install.
apt-get update && apt-get upgrade

apt-get install systemd
apt-get install systemd-sysv
apt-get install ca-certificates
reboot
2. At the end of the install, the script will instruct you to go to the ip address of the server (or domain name) in your
web browser to login. The script will also provide a username and secure random password for you to use. This can
be changed after you login. The install script builds the fusionpbx database. If you need the database password it is
located in /etc/fusionpbx/config.php .
Installation has completed.
Use a web browser to login.

domain name: https://000.000.000.000
username: admin
password: zxP5yatwMxejKXd
The domain name in the browser is used by default as part of the authentication.
If you need to login to a different domain then use username@domain.
username: admin@000.000.000.000
(continues on next page)
2.1. Getting Started 7

(continued from previous page)
Official FusionPBX Training

Admin Training 24 - 26 Jan (3 Days)
Advanced Training 31 Jan - Feb 2 (3 Days)
Continuing Education Last Thursday Monthly (1 Day)
Timezone: https://www.timeanddate.com/worldclock/usa/boise
For more info visit https://www.fusionpbx.com/training.php
Additional information.
https://fusionpbx.com/support.php
https://www.fusionpbx.com
http://docs.fusionpbx.com
https://www.fusionpbx.com/training.php
After the install script has completed go to your web browser and login with the information provided by the install
script.
After the installation script finishes, the option for anything to register to the ip address is ENABLED.
• If you plan on registering devices to the FusionPBX ip address then no further action is required.
It is however recommended to register to a domain name (FQDN) since most scripted attacks happen to the public ip.
Registering to the ip address will be blocked by the fail2ban rules freeswitch-ip and auth-challenge once these rules
are set to true.
• To help secure your FusionPBX installation, enable the fail2ban rules [freeswitch-ip] and [auth-challenge-ip] in
/etc/fail2ban/jail.local.

[freeswitch-ip]
enabled = true
[auth-challenge-ip]
enabled = true
2.1.3 Let’s Encrypt
Let’s Encrypt is one of the most recent and widely used form of free SSL security and supports wildcard DNS. You
can use Let’s Encrypt with your FusionPBX install and WebRTC like Verto Communicator.
2.1.3.1 Dehydrated (Recommended)
FusionPBX has an option to easliy and quickly install SSL with Let’s Encrypt using letsencrypt.sh With this script
you can choose either to request an SSL certificate with wildcard (*.domain.tld) or hostnames (domain.tld).
The letsencrypt.sh will do the following:
• Download dehydrated.
• Request an SSL certificate from Let’s Encrypt.
• Configure NGINX to use the SSL certificate.
• Combine and place SSL certificate in the proper FreeSWITCH directory for using TLS.
• Test and make sure the SSL cert works and outputs if sucessful.
Using letsencrypt.sh
With letsencrypt.sh you have the choice of creating an SSL certificate for a single domain (domain.tld), multiple sub
domains(sub.domain.tld, sub1.domain.tld, etc.domain.tld) or wildcard (*.domain.tld). The easy way however is using
the hostname method.
Hostname
To create a hostname or multiple hostname SSL certificate go to:
cd /usr/src/fusionpbx-install.sh/debian/resources/
Then execute the script.
./letsencrypt.sh
You should then see and follow the prompts.
Domain Name: domain.tld

Email Address: support@fusionpbx.com
After that, you should see the following output.

Cloning into 'dehydrated'...

remote: Counting objects: 1914, done.
remote: Total 1914 (delta 0), reused 0 (delta 0), pack-reused 1914
Receiving objects: 100% (1914/1914), 616.01 KiB | 0 bytes/s, done.
Resolving deltas: 100% (1199/1199), done.
# INFO: Using main config file /etc/dehydrated/config
+ Generating account key...
+ Registering account key with ACME server...
+ Done!
+ Creating chain cache directory /etc/dehydrated/chains
Processing domain.tld
+ Creating new directory /etc/dehydrated/certs/domain.tld ...
+ Signing domains...
+ Generating private key...
+ Generating signing request...
+ Requesting new certificate order from CA...
+ Received 1 authorizations URLs from the CA
+ Handling authorization for domain.tld
+ 1 pending challenge(s)
+ Deploying challenge tokens...
+ Responding to challenge for domain.tld authorization...
+ Challenge is valid!
+ Cleaning challenge tokens...
+ Requesting certificate...
+ Checking certificate...
+ Done!
+ Creating fullchain.pem...
+ Done!
nginx: the configuration file /etc/nginx/nginx.conf syntax is ok

nginx: configuration file /etc/nginx/nginx.conf test is successful
Wildcard
To create a wildcard SSL certificate go to:
Then execute the script.
./letsencrypt.sh
You should then see and follow the prompts:
Domain Name: *.domain.tld

Email Address: support@fusionpbx.com
Cloning into 'dns-01-manual'...

remote: Counting objects: 9, done.
remote: Total 9 (delta 0), reused 0 (delta 0), pack-reused 9
Unpacking objects: 100% (9/9), done.
Checking connectivity... done.
+ Account already registered!


Processing *.domain.tld
+ Checking domain name(s) of existing cert... changed!
+ Domain name(s) are not matching!
+ Names in old certificate: domain.tld
+ Configured names: *.domain.tld
+ Forcing renew.
+ Checking expire date of existing cert...
+ Valid till Nov 19 16:08:32 2018 GMT (Longer than 30 days). Ignoring because renew
˓→was forced!
+ Signing domains...
+ Generating private key...
+ Generating signing request...
+ Requesting new certificate order from CA...
+ Received 1 authorizations URLs from the CA
+ Handling authorization for domain.tld
+ 1 pending challenge(s)
+ Deploying challenge tokens...
Note: When you define the txt record with your domain registrar be sure to use the output of the script you are
running and not what is in this example.
Add the following to the zone definition of domain.tld:

_acme-challenge.domain.tld. IN TXT "PY7ttk6no_5eG7WtAbO6qs5-NzA-Kigko375omKc0nw"
**Press enter to continue...**
+ Responding to challenge for domain.tld authorization...

+ Challenge is valid!
+ Cleaning challenge tokens...
Now you can remove the following from the zone definition of domain.tld:
_acme-challenge.domain.tld. IN TXT "PY7ttk6no_5eG7WtAbO6qs5-NzA-Kigko375omKc0nw"
**Press enter to continue...**
+ Requesting certificate...
+ Checking certificate...
+ Done!
+ Creating fullchain.pem...
deploy_cert()
Done!
**done**
nginx: the configuration file /etc/nginx/nginx.conf syntax is ok

nginx: configuration file /etc/nginx/nginx.conf test is successful
Tip: Use the dig command to check that the txt record is correct. dig -t txt _acme-challenge.domain.tld
Output should show:

;; ANSWER SECTION: _acme-challenge.domain.tld. 1799 IN TXT “PY7ttk6no_5eG7WtAbO6qs5-NzA-

Kigko375omKc0nw”
Setup for multiple domains on Let’s Encrypt
Before setting up multiple domains, make sure you have SSL working on your main domain using the instructions
above.
Create shared nginx host file for all domains
vim /etc/nginx/includes/fusionpbx-default-config
Paste the code below into the file
ssl_protocols TLSv1 TLSv1.1 TLSv1.2;

ssl_ciphers HIGH:!ADH:!MD5:!aNULL;
#letsencrypt
location /.well-known/acme-challenge {
root /var/www/letsencrypt;
}
#REST api
if ($uri ~* ^.*/api/.*$) {
rewrite ^(.*)/api/(.*)$ $1/api/index.php?rewrite_uri=$2 last;
break;
}
#algo
rewrite "^.*/provision/algom([A-Fa-f0-9]{12})\.conf" /app/provision/?mac=$1&file=algom
˓→%7b%24mac%7d.conf last;
#mitel
rewrite "^.*/provision/MN_([A-Fa-f0-9]{12})\.cfg" /app/provision/index.php?mac=$1&
˓→file=MN_%7b%24mac%7d.cfg last;
rewrite "^.*/provision/MN_Generic.cfg" /app/provision/index.php?mac=08000f000000&

˓→file=MN_Generic.cfg last;
#grandstriam
rewrite "^.*/provision/cfg([A-Fa-f0-9]{12})(\.(xml|cfg))?$" /app/provision/?mac=$1;
#aastra
rewrite "^.*/provision/aastra.cfg$" /app/provision/?mac=$1&file=aastra.cfg;
#rewrite "^.*/provision/([A-Fa-f0-9]{12})(\.(cfg))?$" /app/provision/?mac=$1 last;
#yealink common
rewrite "^.*/provision/(y[0-9]{12})(\.cfg)?$" /app/provision/index.php?file=$1.cfg;
#yealink mac
rewrite "^.*/provision/([A-Fa-f0-9]{12})(\.(xml|cfg))?$" /app/provision/index.php?mac=
˓→$1 last;
#polycom
rewrite "^.*/provision/000000000000.cfg$" "/app/provision/?mac=$1&file={%24mac}.cfg";
#rewrite "^.*/provision/sip_330(\.(ld))$" /includes/firmware/sip_330.$2;
rewrite "^.*/provision/features.cfg$" /app/provision/?mac=$1&file=features.cfg;


rewrite "^.*/provision/([A-Fa-f0-9]{12})-sip.cfg$" /app/provision/?mac=$1&file=sip.
˓→cfg;
rewrite "^.*/provision/([A-Fa-f0-9]{12})-phone.cfg$" /app/provision/?mac=$1;

rewrite "^.*/provision/([A-Fa-f0-9]{12})-registration.cfg$" "/app/provision/?mac=$1&
˓→file={%24mac}-registration.cfg";
#cisco
rewrite "^.*/provision/file/(.*\.(xml|cfg))" /app/provision/?file=$1 last;
#Escene
rewrite "^.*/provision/([0-9]{1,11})_Extern.xml$" "/app/provision/?ext=$1&file={
˓→%24mac}_extern.xml" last;
rewrite "^.*/provision/([0-9]{1,11})_Phonebook.xml$" "/app/provision/?ext=$1&file={

˓→%24mac}_phonebook.xml" last;
access_log /var/log/nginx/access.log;
error_log /var/log/nginx/error.log;
client_max_body_size 80M;
client_body_buffer_size 128k;
location / {
root /var/www/fusionpbx;
index index.php;
}
location ~ \.php$ {
fastcgi_pass unix:/var/run/php/php7.1-fpm.sock;
#fastcgi_pass 127.0.0.1:9000;
fastcgi_index index.php;
include fastcgi_params;
fastcgi_param SCRIPT_FILENAME /var/www/fusionpbx$fastcgi_script_name;
}
# Disable viewing .htaccess & .htpassword & .db

location ~ .htaccess {
deny all;
}
location ~ .htpassword {
deny all;
}
location ~^.+.(db)$ {
deny all;
}
Create a file to contain config for additional domains

touch /etc/nginx/includes/fusionpbx-domains
make default file read configs for additional domains
vim /etc/nginx/sites-available/fusionpbx
Add the line below at the very end of the file after the trailing “}”
include /etc/nginx/includes/fusionpbx-domains;
By now you are all set to start using SSL on multiple domains for your FusionPBX installation.
Follow the steps below everytime your add a new domain

Create a conf file for the new domain (repalce example.com with your own domain)
vim /etc/letsencrypt/configs/example.com.conf
Paste this into the .conf file (don’t forget to change the defaults, especially the domain)
# the domain we want to get the cert for;

# technically it's possible to have multiple of this lines, but it only worked
# with one domain for me, another one only got one cert, so I would recommend
# separate config files per domain.
domains = my-domain
# increase key size

rsa-key-size = 2048 # Or 4096
# the current closed beta (as of 2015-Nov-07) is using this server

server = https://acme-v01.api.letsencrypt.org/directory
# this address will receive renewal reminders

email = my-email
# turn off the ncurses UI, we want this to be run as a cronjob

text = True
# authenticate by placing a file in the webroot (under .well-known/acme-

˓→upatechallenge/)
# and then letting LE fetch it

authenticator = webroot
webroot-path = /var/www/letsencrypt/
Obtain the cert from Let’s Encrypt (again, replce example.com with your domain)
cd /opt/letsencrypt
./letsencrypt-auto --config /etc/letsencrypt/configs/example.com.conf certonly
Set cert to auto renew with other domains
cd /etc/fusionpbx
vim renew-letsencrypt.sh
Add the line below right below where it says “cd /opt/letsencrypt/” (again replace example.com with your domain)
./certbot-auto --config /etc/letsencrypt/configs/example.com.conf certonly
--non-interactive --keep-until-expiring --agree-tos --quiet
Finally add your new domain to be loaded
vim /etc/nginx/includes/fusionpbx-domains
Paste the below at the very end of the file (again replace example.com with your domain)
server {
listen 443;
server_name example.com;
ssl on;
ssl_certificate /etc/letsencrypt/live/example.com/fullchain.pem;
ssl_certificate_key /etc/letsencrypt/live/example.com/privkey.pem;
include /etc/nginx/includes/fusionpbx-default-config;
}

You’re all set! Restart nginx for changes to take effect

service nginx restart
2.1.4 Provision
2.1.4.1 Automatic
Auto provisioning is disabled by default. This is to give a chance to secure provisioning server with HTTP Authen-
tication or CIDR. HTTP Authentication requires the phone to send hash of the combined username and password in
order to get configuration. CIDR is an IP address restriction that can be used to restrict which IP addresses are allowed
to get the device configuration. An example of CIDR is xxx.xxx.xxx.xxx/32 the /32 represents a single IP address.
To set one of these values go to Advanced > Default Settings and find the Provision category from there used the edit
button to set a value. After this is done it is safe to set enabled equal to true.
• Yealink
• Polycom
• Cisco
• Fanvil
• Grandstream
• Htek
• Zoiper
2.1.4.2 Manual
How to setup the device using the phone’s web interface.

• Yealink
• Polycom
• Cisco
• Fanvil
• Grandstream
• Htek
• SNOM
• Zoiper
2.1.4.3 Advanced > Default Settings
In the Provisioning section, there are a few key options that have to be set in order to turn auto provisioning on.
• enabled Must be enabled and set to value true and enabled True. It is disabled by default.
• http_auth_username Must be enabled and set to value true and enabled True. It is disabled by default. Be
sure to use a strong username.
• http_auth_password Must be enabled and set to value true and enabled True. It is disabled by default. Be
sure to use a strong password.

• cidr Optional security option to allow configuration request limited to specific IP version 4 ranges. Type array
allows multiple ranges of IP addresses.
2.1.4.4 Phone Screen Capture
• Screen Capture
Note: Click here to view how to add a device.
2.1.4.5 Phone Book
Remote phone book (Address Book) are based on the FusionPBX Contacts App.
Phone Book Settings
In order to use the phone book a few steps are needed.

• Assign the device to a user.
• Create or import the Contacts.
• Set Enabled as True in Default Settings.
• Set Enabled True for contact_extensions, contact_users and contact_groups in Default Settings.
• From the phone, go into the menu to update the phone book.

2.1.5 Security
Similar to medieval fortifications it is recommended to provide your servers with multiple layers of defenses. Be sure
to use Firewalls, Strong passwords, SSH, and make sure your servers are kept up to date for all software being used.
This includes the operating system, FreeSWITCH and FusionPBX.
2.1.5.1 FusionPBX
The latest Debian install script configures IPTables firewall for you. FusionPBX extensions set strong passwords for
you by default. You can increase the password complexity using settings in Advanced -> Default Settings to increase
the length of the passwords that are generated by default.
2.1.5.2 Firewall
Although the new install script configured IPTables for you it is recommended that you review the settings. On Debian
and Ubuntu you can check your firewall with the following command.
iptables -L
• Firewall page
2.1.5.3 SSL / TLS
SSL and TLS are very necessary in today’s internet applications from VOIP to Websites. FusionPBX by default uses a
self signed certificate. However you can use certificate providers where you can purchase certificates and there are free
options as well. With domain based multi-tenant wildcard certificates can be useful. Also when deciding on which
certificate provider to use you should look at the phones manufacturer’s documentation to find one that is compatible
with HTTPS provisioning.
Let’s Encrypt provides free certificates for a single domain and they also now support wildcard certificates.
• Setup Let’s Encrypt with FusionPBX
2.1.5.4 Upgrade
Security problems are fixed as they are discovered and are updated for master and the latest release. Upgrades are
considered an important part of keeping the server secure. Upgrades always need to be done on the operating system,
FreeSWITCH and FusionPBX.
Latest install script will install FreeSWITCH packages by default. To upgrade them and operating system packages
run the following commands.
apt-get update
apt-get upgrade
If you need help upgrading safely please consider paid support.
2.1.5.5 XML RPC
New install mod_xml_rpc is not enabled by default. It is recommended to run a firewall on all FusionPBX servers.
The latest Debian install script configures the firewall by default. However it is recommended to check to make sure it
is installed and running.

Mod_xml_rpc allows running remote commands to FreeSWITCH. Ensure you have a firewall that is protecting the
XML RPC port. Consider changing the XML RPC password. Advanced -> Settings page in the interface allows you
to change the password or the port. At the very least do not allow public access to the XML RPC port.
The latest Debian install script installs iptables firewall which prevents public access to the mod_xml_rpc port. If you
are not using a firewall on the server you should even if its protected by by an external firewall. Some uninformed
co-worker could expose the server to the public internet at some point in the future. Multiple layers of security are
considered best practice.
XML RPC is secure by default for 2 reasons.
• The module is disabled by default.
• Install script firewalls XML RPC port 8787 and does not allow access to it by default outside of 127.0.0.1.
If you were to start the module and open port 8787 on the firewall you would want to set a really good password for
it under Advanced -> Settings. It would be recommended to use a VPN such as OpenVPN to access XML RPC over
port 8787 instead of opening port 8787 on the firewall.
2.1.5.6 Fail2ban
Fail2ban is also used to protect SSH, FreeSWITCH, the web server as well as other services. You can view any IP
addresses that have been blocked by Fail2ban with the following command.
iptables -L
2.1.5.7 SSH
Use strong passwords with SSH or even better use SSH keys for better protection of your servers.
2.1.6 Backup
It’s always good to have a backup method in place. Here are the steps to a basic backup method with FusionPBX. The
install script on Debian will automatically copy this backup script to /etc/cron.daily/fusionpbx-backup. Backups get
stored in /var/backups/fusionpbx/postgresql by default.
2.1.6.1 Command Line
Be sure to change the password by replacing the zzzzzzzz in PGPASSWORD=”zzzzzzzz” with your database pass-
word. You can get the password from /etc/fusionpbx/config.php.
cd /etc/cron.daily
nano fusionpbx-backup
#!/bin/sh
export PGPASSWORD="zzz"
db_host=127.0.0.1
db_port=5432

now=$(date +%Y-%m-%d)
mkdir -p /var/backups/fusionpbx/postgresql
echo "Backup Started"
#delete postgres backups

find /var/backups/fusionpbx/postgresql/fusionpbx_pgsql* -mtime +4 -exec rm {} \;
#delete the main backup

find /var/backups/fusionpbx/*.tgz -mtime +2 -exec rm {} \;
#backup the database

pg_dump --verbose -Fc --host=$db_host --port=$db_port -U fusionpbx fusionpbx --
˓→schema=public -f /var/backups/fusionpbx/postgresql/fusionpbx_pgsql_$now.sql
#package
#tar --exclude='/var/lib/freeswitch/recordings/*/archive' -zvcf /var/backups/
˓→fusionpbx/backup_$now.tgz /var/backups/fusionpbx/postgresql/fusionpbx_pgsql_$now.
˓→sql /var/www/fusionpbx /usr/share/freeswitch/scripts /var/lib/freeswitch/storage /
˓→var/lib/freeswitch/recordings /etc/fusionpbx /etc/freeswitch /usr/share/freeswitch/
˓→sounds/music/
#source
#tar -zvcf /var/backups/fusionpbx/backup_$now.tgz /var/backups/fusionpbx/postgresql/
˓→fusionpbx_pgsql_$now.sql /var/www/fusionpbx /usr/local/freeswitch/scripts /usr/
˓→local/freeswitch/storage /usr/local/freeswitch/recordings /etc/fusionpbx /usr/local/
˓→freeswitch/conf /usr/local/freeswitch/sounds/music/
echo "Backup Completed"
To save the file press ctrl + x then y to save it.

You should have the script ready to execute. (Default the script will use FreeSWITCH package paths. If you have an
older install using source be sure to change this by commenting the package line #22 and uncomment the source line
#25.)
2.1.6.2 Crontab (optional)
Files in /etc/cron.daily will execute automatically if they don’t have an extension like .sh for this reason the backup
script was renamed from fusionpbx-backup.sh to fusionpbx-backup and then it runs nightly without needing to use
crontab.
Setting crontab -e
crontab -e
Choose 1 for nano
Goto the last blank line and paste in the next line.
0 0 * * * /bin/sh /etc/cron.daily/fusionpbx-backup.sh
press enter then save and exit.
Once this is complete you will have the backup ready to execute by ./fusionpbx-backup or from the daily cron job.

2.1.7 Restore
It’s always good to have a restore method of a backup in place. Here are the steps to a basic restore method with
FusionPBX.
Note: It is important to know if your installation is from package or source as the paths are different for FreeSWITCH.
Always test the backups and restore methods on test machines first.
• To create the script use an editor such as vi or nano.

• Copy/Paste from the code block below and save the file as fusionpbx-restore.sh
• Replace zzz with your database password
• chmod + x fusionpbx-restore.sh and then run the script ./fusionpbx-restore.sh
• edit the script as needed and run this script from the server you are restoring on.
#!/bin/sh
now=$(date +%Y-%m-%d)
ssh_server=x.x.x.x
database_host=127.0.0.1
database_port=5432
export PGPASSWORD="zzz"
#run the remote backup

ssh -p 22 root@$ssh_server "nice -n -20 /etc/cron.daily/./fusionpbx-backup.sh"
#delete freeswitch logs older 7 days

find /var/log/freeswitch/freeswitch.log.* -mtime +7 -exec rm {} \;
#synchronize the backup directory

#rsync -avz -e 'ssh -p 22' root@$ssh_server:/var/backups/fusionpbx /var/backups
rsync -avz -e 'ssh -p 22' root@$ssh_server:/var/backups/fusionpbx/postgresql /var/
˓→backups/fusionpbx
rsync -avz -e 'ssh -p 22' root@$ssh_server:/var/www/fusionpbx /var/www

rsync -avz -e 'ssh -p 22' root@$ssh_server:/etc/fusionpbx /etc
find /var/backups/fusionpbx/postgresql -mtime +2 -exec rm {} \;
rsync -avz -e 'ssh -p 22' root@$ssh_server:/etc/freeswitch/ /etc

rsync -avz -e 'ssh -p 22' root@$ssh_server:/var/lib/freeswitch/storage /var/lib/
˓→freeswitch
rsync -avz -e 'ssh -p 22' root@$ssh_server:/var/lib/freeswitch/recordings /var/lib/

˓→freeswitch
rsync -avz -e 'ssh -p 22' root@$ssh_server:/usr/share/freeswitch/scripts /usr/share/

˓→freeswitch
rsync -avz -e 'ssh -p 22' root@$ssh_server:/usr/share/freeswitch/sounds /usr/share/

˓→freeswitch
echo "Restoring the Backup"

#extract the backup from the tgz file
#tar -xvpzf /var/backups/fusionpbx/backup_$now.tgz -C /
#remove the old database



psql --host=$database_host --port=$database_port --username=fusionpbx -c 'drop
˓→schema public cascade;'
psql --host=$database_host --port=$database_port --username=fusionpbx -c 'create

˓→schema public;'
#restore the database

pg_restore -v -Fc --host=$database_host --port=$database_port --dbname=fusionpbx --
˓→username=fusionpbx /var/backups/fusionpbx/postgresql/fusionpbx_pgsql_$now.sql
#restart freeswitch
service freeswitch restart
echo "Restore Complete";
2.1.8 Firewall
Basic ports used

• SIP TCP/UDP
– 5060-5090
• RTP UDP
– 16384-32768
• SSH
– 22
• HTTP
– 80, 443
2.1.8.1 Iptables
Iptables are used in the Debian install script.
Basic Rules
iptables -A INPUT -i lo -j ACCEPT

iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
iptables -A INPUT -p tcp --dport 22 -j ACCEPT
iptables -A INPUT -p tcp --dport 5060:5069 -j ACCEPT
iptables -A INPUT -p udp --dport 5060:5069 -j ACCEPT
iptables -A INPUT -p udp --dport 5080 -j ACCEPT
iptables -A INPUT -p udp --dport 16384:32768 -j ACCEPT
iptables -A INPUT -p icmp --icmp-type echo-request -j ACCEPT
iptables -A INPUT -p udp --dport 1194 -j ACCEPT
iptables -P INPUT DROP
iptables -P FORWARD DROP
iptables -P OUTPUT ACCEPT

Optional Rules
OPENVPN: iptables -A INPUT -p udp --dport 1194 -j ACCEPT

ICMP: iptables -A INPUT -p icmp --icmp-type echo-request -j ACCEPT
Friendly Scanner
Rules to block not so friendly scanner
iptables -I INPUT -j DROP -p tcp --dport 5060 -m string --string

"friendly-scanner" --algo bm
iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
Optional

"VaxSIPUserAgent" --algo bm
"VaxIPUserAgent" --algo bm
"VaxSIPUserAgent" --algo bm
"VaxIPUserAgent" --algo bm

"VaxSIPUserAgent/3.1" --algo bm
Show iptable rules
sudo iptables -L -v

Show line numbers
iptables -L -v --line-numbers
Delete a line
Delete line 2
iptables -D INPUT 2
Flush Out Iptables
iptables -P INPUT ACCEPT

iptables -P FORWARD ACCEPT
iptables -P OUTPUT ACCEPT
iptables -F
Open a Port for a Specific IP Address
iptables -A INPUT -j ACCEPT -p tcp –dport 5432 -s x.x.x.x/32
Block IP address
iptables -I INPUT -s 62.210.245.132 -j DROP
Flush iptables
How to flush iptables without loosing access to ssh.
iptables -P INPUT ACCEPT

iptables -F
Save Changes
Debian / Ubuntu
apt-get install iptables-persistent

service iptables-persistent save
dpkg-reconfigure iptables-persistent

2.1.8.2 Fail2ban
Fail2ban is also used to protect SSH, FreeSWITCH, the web server as well as other services.
After the installation script finishes, the option for anything to register to the ip address is ENABLED.
• If you plan on registering devices to the FusionPBX ip address then no further action is required.
It is however recomended to register to a domain name (FQDN) since most scripted attacks happen to the public ip.
Registering to the ip address will be blocked by the fail2ban rules freeswitch-ip and auth-challenge once these rules
are set to true.
• To help secure your FusionPBX installation, enable the fail2ban rules [freeswitch-ip] and [auth-challenge-ip] in
/etc/fail2ban/jail.local.
[freeswitch-ip]
enabled = true
[auth-challenge-ip]
enabled = true
Warning: If you find that your FusionPBX web interface isn’t loading then check and see if fail2ban is blocking
your ip. Getting blocked by any fail2ban rule will block ssh, www, and phones registering if you don’t have your
ip in the /etc/fail2ban/jail.conf ignoreip= field .
You can view the IP addresses blocked by Fail2ban with the following command.
iptables -L -n
To check the status of one of the fail2ban jails
fail2ban-client status freeswitch-ip
Fail2ban configuration files are located in.
cd /etc/fail2ban/
To exclude an IP so that it isn’t blocked by any filters edit the jails.conf file.
nano /etc/fail2ban/jail.conf
Find ignoreip and add the IP address, CIDR or DNS hostname that need to be white listed. Use a space as a delimitter
between each one. Restart fail2ban to apply the changes to the ignoreip list.
ignoreip = 127.0.0.1/8 192.168.0.0/16
Note: To help keep the ip and hostnames you want unblocked it is a good idea to add customers and carriers to the
ignoreip list.
Filters are defined in the following directory.
/etc/fail2ban/filter.d
Inside jail.local points to filters and defines maxretry, bantime, logpath, ports to block and more.

/etc/fail2ban/jail.local
Clear all blocked addresses by restarting fail2ban.
service fail2ban restart
Fail2ban logs the addresses that it blocks with the filter that triggered it.
/var/log/fail2ban.log
More information about Fail2ban can be found at http://www.fail2ban.org/wiki
Note: You can use a dynamic ip address service like dyndns to whitelist a dynamic ip address.
2.1.8.3 PF
Packet Filter is used in the FreeBSD setup script.
Basic Rules
set skip on lo0

scrub in all
antispoof for lo0

table <fail2ban> persist
pass out quick all

pass quick on lo0 all
block in all
block in quick from <fail2ban>
pass in quick inet proto icmp all
pass in quick inet6 proto icmp6 all
pass in quick inet proto tcp from any to any port 22 keep state
pass in quick inet proto udp from any to any port 5060 keep state
pass in quick inet proto udp from any to any port 5080 keep state
pass in quick inet proto udp from any to any port 16384:32768 keep state

Disable
pfctl -d
Enable
pfctl -e
Show Rules
pfctl -s rules
2.1.9 Languages
FusionPBX has multilingual capabilities. This will allow for different languages to be used in your FusionPBX
installation. Languages can be set globally, per tenant and per user. In addition to your FusionPBX installation web
interface, there are options to upload audio files for FreeSWITCH to use via command line.
2.1.9.1 Fusionpbx Settings
Global
Advanced > Default Settings

Setting the language from here will set the language for the entire FusionPBX installation.
Domain (Tenant)
Advanced > Domains then click the plus at the bottom right and fill in the required fields.
Setting the language from here will set the language for the entire domain (tenant) in your FusionPBX installation.
This can override the Global language settings.

User
Accounts > Users then edit the user.

Setting the language from here will set the language for this specific user and will override Global and Domain
language settings.

2.1.9.2 FreeSWITCH Sound Files
FreeSWITCH sound files location are dependent on operating system and installation method.
Package Install
• Most if not all recent installations of FusionPBX are using packages for FreeSWITCH.
• File system location:
/usr/share/freeswitch/sounds/en/us/
Source Install
• Older installs, custom installs, or personal preference are using source compiled versions.
• File system location:
/usr/local/freeswitch/sounds/en/us/
Where to get language sounds
• Free: https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse
2.1.9.3 app_languages.php
Guidelines The words used in the text variable name

• separated with a dash.
• begin with a prefix
• are lower case
Prefixes
• title: The title of the page
• header: The header of the page
• description: Information to describe the page or an item on the page
• button: The label for the buttons
• confirm: A message used to confirm and action like delete
• message: The response after an action is taken
• label: The label for items on the page
• option: The options in an html select box
Languages
Each word, phrase, or sentence has the language declared with the 2 language code with s dash seperating the region.
There is one difference the region is entirely in lower case. For additional information see the following.
http://www.w3.org/International/articles/language-tags/

http://www.iana.org/assignments/language-subtag-registry
• en-us
• es-mx
• de-ch
• de-at
• fr-ca
• fr-ch
• pt-pt
• pt-br
Example File
An excerpt from the app_languages.php for Conference Center.
<?php
$text['title-conference-center']['en-us'] = 'Conference Center';

$text['title-conference-center']['pt-pt'] = '';
$text['header-conference-center']['en-us'] = 'Conference Center';

$text['header-conference-center']['pt-pt'] = '';
$text['description-conference-center']['en-us'] = 'Conference Center is used

˓→to setup one or more conference rooms with a name, extension number, a required pin
˓→number length, and a description.';
$text['description-conference-center']['pt-pt'] = '';
$text['label-name']['en-us'] = 'Name';
$text['label-name']['pt-pt'] = '';
$text['label-extension']['en-us'] = 'Extension';
$text['label-extension']['pt-pt'] = '';
$text['label-delete']['en-us'] = 'Delete';
$text['label-delete']['pt-pt'] = '';
$text['label-edit']['en-us'] = 'Edit';
$text['label-edit']['pt-pt'] = '';
$text['button-view']['en-us'] = 'View';
$text['button-view']['pt-pt'] = '';
$text['button-back']['en-us'] = 'Back';
$text['button-back']['pt-pt'] = 'Voltar';
$text['confirm-update']['en-us'] = 'Update Complete';

$text['confirm-update']['pt-pt'] = 'Actualização Completa';
$text['confirm-delete']['en-us'] = 'Do you really want to delete this?';

$text['confirm-delete']['pt-pt'] = '';
$text['button-add']['en-us'] = 'Add';
$text['button-add']['pt-pt'] = '';


$text['button-save']['en-us'] = 'Save';
$text['button-save']['pt-pt'] = 'Guardar';
?>
To use inside the code on each page that displays text. Place the following code at the top just after the permision_exists
//add multi-lingual support

require_once "app_languages.php";
foreach($text as $key => $value) {
$text[$key] = $value[$_SESSION['domain']['language']['code']];
}
To place a word, phrase or sentence it would be used in the code like the following example.
echo "<td align='left' width='30%' nowrap='nowrap'><b>".$text['title-conference-

˓→centers']."</b></td>\n";
An additional example.
echo " <tr>\n";

echo " <td align='left' colspan='2'>\n";
echo " ".$text['description-conference-centers']."\n";
echo " </td>\n";
echo " </tr>\n";
echo "</table>\n";

CHAPTER 3
Home
3.1 Home
The Home menu gives access to Account Settings, Dashboard and the option to Logout.
31
3.1.1 Account Settings
• User Name: The user name.

• Password: The password.
• Confirm Password: Must match the password.
• Language: Choose a language for the user.
• Time Zone: Time zone specific to the user.
• Status: Used for call center and operator panel.
• Contact: The users contact. Is used in a phone directory or Apps > Contacts.
• Groups: Group the user is in and relates to what the user can see and do in the menus.
• Domain: Domain specific to the user.
• Enabled: Enable or disable the account.
3.1.2 Dashboard
Quickly access information and tools related to your account. Depending on the user permissions, the user may see
less options on this screen.
32 Chapter 3. Home
• Voicemail: New and total voicemails related to the users voicemail box. A user can be assigned to more than 1
voicemail box.
• Missed Calls: Missed calls for the user.
• Recent Calls: Number of calls in the last 24 hours.
• System Status: Disk usage in percentage, FusionPBX version, FreeSWITCH version, FreeSWITCH uptime,
OS Uptime, CPU Usage, DB Connections, Channels and Registrations.
• Call Routing: See if call forward, follow me, do not disturb is set and a quick wat to edit those options if
needed.
• Ring Group Forward: See the name, extension number, if forwarding is enabled and what number it is for-
warded to.
• System Counts: Number of Domains, Devices, Extensions, Gateways, Users, Destinations, CC Queues, IVR
Menus, Ring Groups, Voicemail and if they are disabled.
3.1.2.1 Dashboard Default Settings
3.1.3 Logout
Logout when you are done, after an upgrade or specific setting change that requires a new session.
3.1. Home 33
34 Chapter 3. Home
CHAPTER 4
Accounts
4.1 Accounts
In the Accounts menu you have access to devices, extensions, gateways, providers and users.
4.1.1 Devices
Used to define the information needed to assign SIP accounts and keys to provision the devices.
• Click the plus icon to add a device.
• Click the edit pencil icon to edit a device.
• Enter the mac address of the phone.

• Add a label.
• Select from the drop down box the make/model.
• Populate the lines section.
• Populate the Key section.
35
• (Optional) Populate the Settings section. These settings are the same as the variables from Advanced > Default
Settings > Provisioning and can be overridden in this settings section. Just set the variable for the device you
are adding.
• Edit other fields as needed.
• Click Save
• To view steps on how to configure other devices to provision click here for the provisioning section.
4.1.1.1 Device Vendors
Vendors can be added or removed to help fine tune the devices page when configuring specific vendor phones.
36 Chapter 4. Accounts
4.1.1.2 Profiles
Define a set of keys as a profile. Any changes to the profile effect all devices assigned to the profile.
4.1.2 Extensions
Extensions define the information needed for an endpoint such as a hard phone, soft phone or some other device to
connect to the SIP server. The extension is the SIP username and the password is the secret used for authentication.
The domain name servers (DNS) to purposes it, locates the server to register to and is the realm that determines which
domain the endpoint is registering to.
4.1. Accounts 37
4.1.2.1 Basic Settings
• Extension Enter the alphanumeric extension. The default configuration allows 2 - 7 digit extensions.
• Number Alias If the extension is numeric then number alias is optional. The primary purpose of this field
is when the extension is not a number then the number alias is required. Note a numeric extension and
number alias does not currently work.
• Range Enter the number of extensions to create. Increments each extension by 1.
• Voicemail Password Enter the numeric voicemail password here.
• Account Code Used with billing systems if you don’t have a billing system then its optional.
• Effective caller ID Name Internal Caller ID name
• Effective Caller ID Number Internal caller ID number usually set to the extension number.
• Outbound Caller ID Name Used by the outbound route for external caller ID name. Business or Organization
typically is set here.
• Outbound Caller ID Number Used by the outbound route for external caller ID number here. Business or
Organization number goes here.
• Emergency Caller ID Name This is used when calling out to an emergency service like 911.
• Emergency Caller ID Number This is used when calling out to an emergency service like 911.
• Directory Full Name The first and last name used in the directory. You can call that directory with *411
• Directory Visible Select whether to hide the name from the directory.
• Directory Extension Visible Select whether announce the extension when calling the directory.
• Limit Max Set max number of outgoing calls for this user.
• Limit Destination Set the destination to send the calls when the max number of outgoing calls has been
reached.
• Voicemail Enabled Enable or disable voicemail for this extension.
• Voicemail Mail To The email address for sending voicemail to email.
• Voicemail File Select whether to send the voicemail as an attachment or as a link in the email.
• Voicemail Keep Local Choose whether to keep the voicemail in the system after sending the email notification.
• Missed Call Set the missed call to true and set the email address if you want to receive an email for missed
calls that were routed through the dialplan to and was not answered by the extension.
• Toll Allow Enter the toll allow value here. (Examples: domestic,international,local) This can be set to any
name you want it sets a variable that can be a condition on the outbound routes.
• Call Timeout Set the timeout for the call ringing.
• Call Group You can define any call group you want the following groups are examples: sales, support, billing.
These are used for group intercept or calls can be sent to the call group.
• Call Screen Call screen if set will ask the caller to identify themselves their response will be recorded and
offered to the person reciving the call.
• Record Whether to record local, inbound, outbound, or all calls that were sent directly to this extension.
• Hold Music Select music or ring tones that will be used for music on hold for this extension.
• Context The context is set by default to match the domain name or IP addres. It is usually correct by default
and doesn’t need to be changed in most cases.
• Enabled Extension enabled or disabled.
• Description A description for the extension.
4.1.2.2 Advanced Settings
Advanced settings in extensions. Be sure to know what and why you are changing these settings or you will risk
causing issues for the extention.
• Auth ACL Advanced auth acl uses.

• CIDR Advanced cidr uses.
• SIP Force Contact Choose whether to rewrite the contact port, or rewrite both the contact IP and port.
• SIP Force Expires To prevent stale registrations SIP Force expires can override the client expire.
• MWI Account MWI Account with user@domain of the voicemail to monitor.
• SIP Bypass Media Choose whether to send the media stream point to point or in transparent proxy mode.
• Absolute Codec String Absolute Codec String for the extension.
• Force ping Use OPTIONS to detect if extension is reacheable.
• Domain The domain the extension is currently saved on.
• Dial String Location of the endpoint.
4.1.3 Gateways
Gateways define the location and settings for other VoIP servers or Providers. After defining the Gateways use the
Outbound routes to direct calls through the gateways. Required items are in bold. Its a good idea to start with the
required items test it and then make adjustments as needed.
4.1. Accounts 39
Gateways provide access into other voice networks. These can be voice providers or other systems that require SIP
registration. Check out the Youtube video .
In this example we will be using VoiceTel . Each Gateway provider has their own setings to use.
Click to visit
Select Accounts from the drop-down list and click on Gateways.
Click the
button on the right. Enter the gateway information below and Click on Save once complete.
Gateway: VoiceTel
Username: 0123456789
Password: 1b3d5f7h9j
From user: 0123456789
From domain: sbc.voicetel.com
Proxy: sbc.voicetel.com
Register: true
Enabled: true
4.1.3.1 Basic Settings
• Gateway: The name of the Gateway. The company name or domain name of th VoIP provider is commonly
used for the name.
• Username: This is the username for SIP registration provided by the carrier.
• Password: This is the password for SIP registrations it is provided by the carrier.
• From User: Optional: Set a specific SIP From User
• From Domain: Optional: Sets a specific SIP From Domain.
• Proxy: Required: Proxy server address used by the carrier. This will vary by carrier.
• Realm: Optional: Required by some carriers
4.1. Accounts 41
• Expire Seconds: Optional: The time until the registration with carrier expires.
• Register: Required: Set to true if the carrier uses a username and password. Set to false if the carrier uses IP
authentication. If false, you will need to specify all of the carrier IP’s in the Advanced > Access Controls.
• Context: Required: Default is set to public and usually the correct value.
• Profile: Required: The SIP profile used by default external is used. If you disable the external profile make sure
to change the SIP profile to one that is enabled.
• Hostname: This should usually be left empty. When the hostname is set the gateway will only start on the
matching server with same hostname. If the hostname is left blank the gateway will start regardless of the
server’s hostname.
• Enabled: Required: If the gateway is enabled or disabled.
• Description: It is helpful to provide a good description for the gateway.
4.1.3.2 Advanced Settings
Most settings in the Advanced Gateway Settings can remain the same. Some carriers will require slight changes in
this section to help with outbound caller ID.
• Distinct To:
• Auth Username:
• Extension: Usually used for testing and not for production. Hard codes a set number and all calls would be
hard coded to that number for inbound calls from that gateway.
• Register Transport: Tells the switch to use SIP with TCP, UDP or TLS.
• Register Proxy: Enter the hostname or IP address of the register proxy. host[:port].
• Outbound Proxy: Enter the hostname or IP address of the outbound proxy. host[:port].
• Caller ID In From: If you caller ID isn’t working setting this to true will often fix the problem.
• Supress CNG: Set this value to true to disable comfort noise.
• Sip CID Type: The SIP caller id type: none, pid, and rpid.
• Codec Preferences: Enter the codec preferences as a list. Ex: PCMA,PCMU,G722,OPUS
• Extension In Contact: Option to set the Extension In Contact.
• Ping: If your server is behind NAT then the ping option can be used to keep the connection alive through the
firewall. The ping interval is in seconds.
• Domain: If the gateway will be used on a specific domain or global to all tenants.
Note: To see which Gateway a call is using. Advanced > Command and in the switch command section type show
channels as xml and then press the execute button. In the output that is returned, look for the string sofia/gateway/ and
the gateway name. This is the gateway your call is using.
4.1.4 Users
Define the users information to login to the web interface.

• Username User id to be used to login.
• Password Secret password used to login.
• Language Per user language to override the domain or global language.
• Time Zone Per user time zone only needed if it needs to be different from the global time zone.
• Status Set the user’s presence.
• Contact Assign a contact to this user account. View
• Groups The group the user is assigned.
• Domain The domain the user is assigned to.
4.1. Accounts 43
• API Key Generates an API Key

• Message Key Generates a Key to use with Messages Application.
• Enable Whether the user is enabled.
4.1.4.1 Users Default Settings
Click the link above for Users default settings.
CHAPTER 5
Dialplans
5.1 Dialplans
In the Dialplan menu you have access to Destinations, Dialplan Manager, Inbound Routes and Outbound Routes.
5.1.1 Destinations
Inbound destinations are the DID/DDI, DNIS or Alias for inbound calls. Click here for the youtube video
Configure Inbound Destinations: (This will auto-configure an Inbound Route also)
Tip: Outbound destinations can be created also.
Select Dialplan from the drop-down list and then click Destinations.
To add a destination click on the plus button on the right.
Enter the route information below and Click Save once complete.
45
• Type: Inbound or Outbound. Choose if this is an inbound destination or outbound destination.

• Destination: This is usually the DID a caller will call.
• Context: This will usually be public.
• Actions: Choose where the call will go after it enters FusionPBX.
– Dialplans can also be used as an action. To enable a dialplan to be visable go to Dialplan > Dialplan
Manager and edit a dialplan. Select True from the Destination field and click save. This applies to
dialplans that have a value in the Number field.
• Caller ID Name Prefix: Adds a name to the Caller ID that will display to the endpoint and call detail records.
• Record: Record all calls made to the destination.
• Account Code: Used in some billing systems.
• Domain: The domain can be global to all domains or domain specific.
• Enabled: Enabled will enable the destination or Disabled to disable the destination.
• Description: A way to label and organize what the destination is for.
• Inbound Routes
46 Chapter 5. Dialplans
– Once a Destination is created an inbound route is also created. Click here to view more about Inbound
routes.
Note: Optional: Replace ^(?:+?1)?(d{10})$ in Inbound Routes with either 0123456789 or a DID Number depending
on the Route Destination setting.
5.1.1.1 Destinations Default Settings
5.1.2 Dialplan Manager
The dialplan is used to setup call destinations based on conditions and context. You can use the dialplan to send calls
to gateways, auto attendants, external numbers, to scripts, or any destination.
Dialplan Name Dialplan Number

caller-details
• Details about the caller.
not-found:
• Used to help trigger fail2ban from bogus calls.
call-limit:
• Limit calls based on number of calls and more.
speed_dial: *0[ext]
• Uses LUA for extension speed dial.
agent_status: *22
• Agent login to call center.
page-extension: *8[ext]
• Password protected paging of an extension.
eavesdrop: *33[ext]
• Password protected evesdropping on extensions.
send_to_voicemail: *99[ext]
• Sending an active call to an extensions voicemail.
cf: cf
•
echo: *9196
• Real time echo test.
milliwatt: *9197
• Plays a milliwatt test tone.
recordings: *732
• Password protected way to record audio that can be used in
other applications like IVR.
directory: *411
• Directory of users.
user_exists:
• Determines if a user exists on the switch.
caller-details:
• Logic to decifer caller details.
call-direction:
• Determines the direction of the call.
variables:
• Set variables on a domain level.
Continued on next page
5.1. Dialplans 47
Table 1 – continued from previous page

is_local:
• Can be used to evaluate calls as local.
call_block:
• Block calls from reaching endpoints.
user_record:
• Used to record calls.
redial: *870
• Dial the last number that was dialed.
default_caller_id:
• Caller ID that can be set per domain.
agent_status_id: *23
• Status of the agent.
provision: *11,*12
• Used with devices.
clear_sip_auto_answer:
•
nway_conference nway
•
cidlookup:
•
group-intercept: *8
• Intercepts a call from a defined group.
page: *724
• Password protected paging defined set of extensions.
conf-xfer:
•
call_privacy: *67[d+]
• Send a privacy header to the carrier to hide caller id.
call_return: *69
• Call the last number that called the endpoint.
extension_queue: *800[ext]
•
intercept-ext: **[ext]
• Password protected intercept of an extension.
dx: dx
• Direct transfer.
att_xfer: att_xfer
• Attended transfer.
extension-to-voicemail: [ext]
• Used for extension to voicemail.
vmain *98
• Main menu to access any voicemail using a pin number.
xfer_vm xfer_vm
• Transfer to voicemail.
is_transfer is_transfer
• Used for call transfering.
vmain_user *97
• Endpoint’s voicemail using a pin number.
delay_echo *9195
Continued on next page
Table 1 – continued from previous page

• Play back an echo with a 5 second delay.
please_hold
• Plays an audio file when on hold.
is_zrtp_secure
•
is_secure is_secure
•
tone_stream *9198
• tones that stream and sound like Tetris music.
hold_music *9664
• Play music on hold. Good for testing on an endpoint.
freeswitch_conference *9888
• An easy way to join the Cluecon Weekly call.
disa *3472
• Call in to a phone number and provide a pin to dial out.
wake-up *925
• Schedule date and time for an automated call.
extension_queue
•
valet_park park+*5901-*5999
• Default range to valet park calls.
valet_park_in park+*5900
• Default number to send valet calls to.
valet_park_out park+*5901-*5999
• Default range to retreive valet parked calls.
operator 0
• Configurable option for an operator.
operator-forward *000
• Uses dial_string.lua.
do-not-disturb *77,*78,*79
• Turn on, toggle, turn off do not disturb.
call-forward *72,*73,*74
• Turn on, toggle on/off and turn off call forwarding.
follow-me *21
• Forwards call to defined list of phone numbers or extensions.
bind_digit_action
•
call_screen [ext]
• Play an audio file and give options to the caller to record a
short message for the call recipient. Call recipient can then accept or reject the call.
local_extension [ext]
• Examines to see if the extension is local.
voicemail [ext]
• Voicemail for extensions.
5.1. Dialplans 49
5.1.3 Dialplan Details
5.1.3.1 Global
Global specific dialplans are global to all tennants(domains). These can be changed, however the changes apply to all
tennants.
Not Found
Dialplan Dialplan Dialplan Detail Data Dialplan Dialplan Dialplan Dialplan

Detail Detail Detail Detail Detail Detail
Tag Type Break Inline Group Order
condition 0 5
action set call_direction=inbound TRUE 0 10
action log [inbound routes] 404 not TRUE 0 15
found ${sip_network_ip}
Call Forward All

condition ${user_exists} TRUE 0 5
condition ${for- TRUE 0 10
ward_all_enabled}
action transfer ${for- 0 15
ward_all_destination}
XML ${domain_name}
Intercept Ext Polycom
Dialplan Dialplan Dialplan Dialplan De- Dialplan De- Dialplan De- Dialplan De-
Detail Tag Detail Type Detail Data tail Break tail Inline tail Group tail Order
condition destina- ^*97(d+)$ 0 5
tion_number
action answer 0 10
action lua intercept.lua 0 15
$1
Talking Clock Date

condition destina- ^*9171$ 0 5
tion_number
action answer 0 10
action sleep 1000 0 15
action say ${default_language} CUR- 0 20
RENT_DATE pronounced
${strepoch()}
action hangup 0 25
Talking Clock Date And Time

tion_number
action answer 0 10
RENT_DATE_TIME pronounced
${strepoch()}
action hangup 0 25
5.1. Dialplans 51
Outbound Route Example

condition ${user_exists}FALSE 0 0
condition destina- ^+?1?(d{10})$ 0 10
tion_number
action set sip_h_X- 0 20
accountcode=${accountcode}
action export call_direction=outbound 0 30
action unset call_timeout 0 40
action set hangup_after_bridge=true 0 50
action set effec- 0 60
tive_caller_id_name=${outbound_caller_id_name}
action set effec- 0 70
tive_caller_id_number=${outbound_caller_id_number}
action set inherit_codec=true 0 80
action set ignore_display_updates=true 0 90
action set callee_id_number=$1 0 100
action set continue_on_fail=true 0 110
action bridge sofia/gateway/72d236fb- 0 120
945b-4c86-8e75-
af7c6bcf2862/$1
action bridge sofia/gateway/72d236fb- 0 130
945b-4c86-8e75-
af7c6bcf2862/$1
Talking Clock Time

tion_number
action answer 0 10
RENT_TIME pronounced
${strepoch()}
action hangup 0 25
5.1.3.2 Domain Specific
Domain specific dialplans are all the same initially but can be changed. Those changes are per domain, thus helps
FusionPBX acheive multitennancy.
Hold Music

tion_number
condition ${sip_has_crypto}
^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$
0 10
action answer 0 15
action exe- is_secure XML ${context} 0 20
cute_extension
action playback $${hold_music} 0 25
anti- set zrtp_secure_media=true 0 30
action
anti- answer 0 35
action
anti- playback silence_stream://2000 0 40
action
anti- exe- is_zrtp_secure XML ${context} 0 45
action cute_extension
anti- playback $${hold_music} 0 50
action
Agent Status
Dialplan Dialplan Dialplan Detail Dialplan De- Dialplan Dialplan De- Dialplan
Detail Tag Detail Type Data tail Break Detail Inline tail Group Detail Order
tion_number
action set agent_id=${sip_from_user} 0 10
action lua app.lua 0 15
agent_status
5.1. Dialplans 53
Agent Status ID
tion_number
action set agent_id= 0 10
agent_status
DISA
Dialplan Dialplan Dialplan Detail Dialplan Dialplan Dialplan De- Dialplan

Detail Tag Detail Type Data Detail Detail Inline tail Group Detail Order
Break
condition destina- ^*(3472)$ 0 5
tion_number
action answer 0 10
action set pin_number=36227215 0 15
action set di- 0 20
alplan_context=${context}
action lua disa.lua 0 25
Provision
condition destina- ^*11$ on-true 0 5
tion_number
action set reboot=true 0 10
action set action=login 0 15
action lua app.lua pro- 0 20
vision
tion_number
action set reboot=true 1 35
action set ac- 1 40
tion=logout
action lua app.lua pro- 1 45
vision
Call Forward

tion_number
action set request_id=false 0 10
action set enabled=true 0 15
action lua call_forward.lua 0 20
tion_number
action set enabled=false 1 40
tion_number
action set enabled=toggle 2 65
condition destina- ^for- on-true 3 80
tion_number ward+(Q${caller_id_number}E)(?:/(d+))?$
action set for- 3 90
ward_all_destination=$2
Call Block
condition ${call_direction}înbound$ 0 5
call_block
5.1. Dialplans 55
Do Not Disturb
Dialplan Dialplan Dialplan Detail Dialplan Dialplan Dialplan De- Dialplan

Detail Tag Detail Type Data Detail Break Detail Inline tail Group Detail Order
tion_number
action lua do_not_disturb.lua 0 15
condition destina- ^*78$|*363$ on-true 1 25
tion_number
action set enabled=true 1 30
tion_number
action set enabled=false 2 50
condition destina- ^dnd+${caller_id_number}$
on-true 3 65
tion_number
Voicemail(Vmain User)
Dialplan Dialplan Dialplan Detail Dialplan Dialplan Dialplan Dialplan

Detail Tag Detail Type Data Detail Detail Detail Detail
Break Inline Group Order
tion_number
action answer 0 10
action set voice- 0 20
mail_action=check
mail_id=${caller_id_number}
mail_profile=default
action lua app.lua voicemail 0 35
Vmain

condition destina- ^vmain$|^*4000$|^*98$
never 0 5
tion_number
action answer 0 10
mail_action=check
condition destina- ^(vmain$|^*4000$|^*98)(d{2,12})$ 1 40
tion_number
action answer 1 45
mail_action=check
action set voicemail_id=$2 1 60
mail_authorized=false
Directory
tion_number
action lua directory.lua 0 10
Follow Me
tion_number
action answer 0 10
action lua fol- 0 15
low_me.lua
5.1. Dialplans 57
Recordings
tion_number
action answer 0 10
action set record- 0 20
ing_slots=true
action set record- 0 25
ing_prefix=recording
action lua recordings.lua 0 30
Call Privacy
condition destina- ^*67(d+)$ 0 5
tion_number
action privacy full 0 10
action set sip_h_Privacy=id 0 15
action set privacy=yes 0 20
action transfer $1 XML 0 25
${context}
Page

tion_number
action set caller_id_name=Page 0 10
action set caller_id_number= 0 15
action set destinations=101-103,105 0 25
action set moderator=false 0 30
action set mute=true 0 35
action set set 0 40
api_hangup_hook=conference
page-${destination_number}
kick all
action lua page.lua 0 45
Valet Park In

Detail Tag Detail Detail Detail Detail Detail
Type Break Inline Group Order
condition destina- ^(park+)?(*5900)$ 0 5
tion_number
action valet_park park@${domain_name} 0 10
auto in 5901 5999
Valet Park Out
condition destina- ^(park+)?*(59[0- 0 5
tion_number 9][0-9])$
action answer 0 10
action valet_park park@${domain_name} 0 15
$2
5.1. Dialplans 59
Valet Parking
Di- Dialplan Detail Dialplan Detail Data Dialplan Dialplan Dialplan Dialplan
alplan Type Detail Detail Detail Detail
Detail Break Inline Group Order
Tag
condi- destination_number ^(park+)?(*59[0-9][0-9])$ never 0 5
tion
condi- ${sip_h_Referred- sip:(.*)@.* never 0 10
tion By}
action set referred_by_user=$1 0 15
condi- destination_number ^(park+)?(*59[0-9][0-9])$ never 1 25
tion
action set park_in_use=false TRUE 1 30
action set park_lot=$2 TRUE 1 35
condi- destination_number ^(park+)?(*59[0-9][0-9])$ 2 45
tion
condi- ${cond TRUE never 2 50
tion ${sip_h_Referred-
By} == ‘’ ? false :
true}
action set park_in_use=${regex TRUE 2 55
${valet_info
park@${domain_name}}|${park_lot}}
condi- ${park_in_use} TRUE never 3 65
tion
action transfer ${referred_by_user} XML 3 70
${context}
anti- set valet_parking_timeout=180 3 75
action
anti- set valet_hold_music=${hold_music} 3 80
action
anti- set valet_parking_orbit_exten=${referred_by_user} 3 85
action
anti- valet_park park@${domain_name} 3 90
action ${park_lot}
User Exists
Di- Di- Dialplan Detail Data Di- Di- Di- Di-

alplan alplan alplan alplan alplan alplan
condi- 0 5
tion
action set user_exists=${user_exists id ${destina- TRUE 0 10
tion_number} ${domain_name}}
condi- ${user_exists}
^true$ 1 20
tion
action set extension_uuid=${user_data ${destina- TRUE 1 25
tion_number}@${domain_name} var exten-
sion_uuid}
action set hold_music=${user_data ${destina- TRUE 1 30
tion_number}@${domain_name} var
hold_music}
action set forward_all_enabled=${user_data ${desti- TRUE 1 35
nation_number}@${domain_name} var for-
ward_all_enabled}
action set forward_all_destination=${user_data ${des- TRUE 1 40
tination_number}@${domain_name} var
forward_all_destination}
action set forward_busy_enabled=${user_data ${des- TRUE 1 45
forward_busy_enabled}
action set forward_busy_destination=${user_data ${des- TRUE 1 50
forward_busy_destination}
action set forward_no_answer_enabled=${user_data TRUE 1 55
${destination_number}@${domain_name} var
forward_no_answer_enabled}
action set forward_no_answer_destination=${user_data TRUE 1 60
forward_no_answer_destination}
action set forward_user_not_registered_enabled=${user_data TRUE 1 65
forward_user_not_registered_enabled}
action set forward_user_not_registered_destination=${user_data TRUE 1 70
forward_user_not_registered_destination}
action set do_not_disturb=${user_data ${destina- TRUE 1 75
do_not_disturb}
action set call_timeout=${user_data ${destina- TRUE 1 80
call_timeout}
action set missed_call_app=${user_data ${desti- TRUE 1 85
nation_number}@${domain_name} var
missed_call_app}
action set missed_call_data=${user_data ${desti- TRUE 1 90
nation_number}@${domain_name} var
missed_call_data}
action
5.1. set
Dialplans toll_allow=${user_data ${destina- TRUE 1 95 61
toll_allow}
action set call_screen_enabled=${user_data ${des- TRUE 1 100
Caller Details

Detail Tag Detail Detail Detail Detail Detail
Type Break Inline Group Order
condition never 0 5
action set caller_destination=${destination_number}TRUE 0 10
action set caller_id_name=${caller_id_name} TRUE 0 15
action set caller_id_number=${caller_id_number} TRUE 0 20
Call Direction
condition ${call_direction}
^(in- never 0 5
bound|outbound|local)$
anti-action export call_direction=local 0 10
Variables

condition 0 5
action export origina- 0 10
tion_callee_id_name=${destination_number}
action set RFC2822_DATE=${strftime(%a, 0 15
%d %b %Y %T %z)}
Call Limit

condition ${call_direction}
^(inbound|outbound)$ 0 5
action limit hash inbound ${do- 0 10
main_uuid} ${max_calls}
!USER_BUSY
Is Local
condition ${user_exists} FALSE 0 5
is_local
User Record
Dialplan Detail Tag Dialplan Detail Type Dialplan Detail Data

condition
action set user_record=${user_data ${destination_number}@${domain_name} var user_record
action set from_user_exists=${user_exists id ${sip_from_user} ${sip_from_host}}
condition ${user_exists} ^true$
condition ${user_record} âll$
action set record_session=true
condition ${call_direction} înbound$
condition ${user_record} înbound$
condition ${call_direction} ôutbound$
condition ${user_record} ôutbound$
condition ${call_direction} ^local$
condition ${user_record} ^local$
condition ${from_user_exists} ^true$
action set from_user_record=${user_data ${sip_from_user}@${sip_from_host} var user_recor
condition ${from_user_record} âll$
condition ${call_direction} înbound$
condition ${from_user_record} înbound$
condition ${call_direction} ôutbound$
condition ${from_user_record} ôutbound$
condition ${call_direction} ^local$
condition ${from_user_record} ^local$
condition ${record_session} ^true$
action set record_path=${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftim
5.1. Dialplans 63
Table 2 – continued fr
action set record_name=${uuid}.${record_ext}
action set recording_follow_transfer=true
action set record_append=true
action set record_in_progress=true
action record_session ${record_path}/${record_name}
Redial

condi- destina- ^(redial|*870)$ on-true 0 5
tion tion_number
action transfer ${hash(select/${domain_name}- 0 10
last_dial/${caller_id_number})}
condi- never 1 20
tion
action hash insert/${domain_name}- 1 25
last_dial/${caller_id_number}/${destination_number}
Speed Dial
condition destina- ^*0(.*)$ 0 5
tion_number
speed_dial
$1
Default Caller ID
Dialplan Dialplan De- Dialplan Detail Data Dialplan Dialplan Dialplan Dialplan
Detail tail Type Detail Detail Detail Detail
Tag Break Inline Group Order
condi- ${emer- ^$ never 0 5
tion gency_caller_id_number}
action set emer- TRUE 0 10
gency_caller_id_name=${default_emergency_caller_id_name}
action set emer- TRUE 0 15
gency_caller_id_number=${default_emergency_caller_id_number}
condi- ${out- ^$ never 1 25
tion bound_caller_id_number}
action set out- TRUE 1 30
bound_caller_id_name=${default_outbound_caller_id_name}
action set out- TRUE 1 35
bound_caller_id_number=${default_outbound_caller_id_number}
Group Intercept
Dialplan Dialplan Detail Dialplan Dialplan De- Dialplan Dialplan De- Dialplan
Detail Tag Type Detail Data tail Break Detail Inline tail Group Detail Order
tion_number
condition ${sip_h_X- ^(.+)$ on-true 0 10
intercept_uuid}
action intercept $1 0 15
condition 1 25
action answer 1 30
action lua inter- 1 35
cept_group.lua
Conf Xfer

condition destination_number ^conf_add_begin$
action set api_result=${conference(${conf_xfer_number} unmute ${conference_member_id}
action bind_digit_action conf-xfer,*0,api:lua,transfer2.lua ${uuid} conf_enter_number::XML::conf-xfer@${
action bind_digit_action conf-xfer,##,api:lua,transfer2.lua ${uuid} conf_enter_number::XML::conf-xfer@${
action bind_digit_action conf-xfer,*#,api:lua,transfer2.lua ${uuid} conf_add_end::XML::conf-xfer@${doma
action bind_digit_action conf,*#,exec:execute_extension,conf_add_begin XML conf-xfer@${domain_name}
action bind_digit_action none,NONE,api:sleep,1
action set continue_on_fail=true
action transfer conf_enter_number XML conf-xfer@${domain_name}
condition destination_number ^conf_add_end$
action digit_action_set_realm conf
action set api_result=${conference(${conf_xfer_number} mute ${conference_member_id})}
5.1. Dialplans 65
Ta
action conference ${conf_xfer_number}@page
condition destination_number ^conf_enter_number$
action digit_action_set_realm none
action read 2 11 ‘tone_stream://%(10000,0,350,440)’ target_num 30000 #
action execute_extension conf_bridge_${target_num} XML conf-xfer@${domain_name}
condition destination_number ^conf_bridge_$
action execute_extension conf_add_end XML conf-xfer@${domain_name}
condition destination_number ^conf_bridge_*$
action execute_extension conf_add_end XML conf-xfer@${domain_name}
condition destination_number ^conf_bridge_(d{2,7})$
action digit_action_set_realm conf-xfer
action bridge {conf_xfer_number=${conf_xfer_number},transfer_after_bridge=conf_enter_to:XM
action execute_extension conf_enter_number XML conf-xfer@${domain_name}
condition destination_number ^conf_bridge_
action playback voicemail/vm-that_was_an_invalid_ext.wav
action execute_extension conf_enter_number XML conf-xfer@${domain_name}
condition destination_number ^conf_enter_to$
action unbind_meta_app
action bind_digit_action conf,*#,exec:execute_extension,conf_add_begin XML conf-xfer@${domain_name}
action digit_action_set_realm conf
action answer
action playback tone_stream://L=1;%(500, 0, 640)
action conference ${conf_xfer_number}@page
condition destination_number ^conf_xfer_from_dialplan$
action lua transfer2.lua ${uuid} conf_add_begin::XML::conf-xfer@${domain_name} conf_en
Page Extension
Dialplan Dialplan Dialplan De- Dialplan De- Dialplan De- Dialplan De- Dialplan De-
Detail Tag Detail Type tail Data tail Break tail Inline tail Group tail Order
condition destina- ^*8(d{2,7})$ 0 5
tion_number
action set destina- 0 10
tions=$1
action set mute=true 0 20
action set modera- 0 25
tor=false
action lua page.lua 0 30
Eavesdrop
tion_number
action answer 0 10
action lua eavesdrop.lua 0 20
$1
Call Return
Dialplan Dialplan Dialplan DetailDdata Dialplan Dialplan Dialplan Dialplan

tion_number
action transfer ${hash(select/${domain_name}- 0 10
call_return/${caller_id_number})}
Extension Queue

condition destina- ^*800(.*)$ 0 5
tion_number
action set fifo_music=$${hold_music} 0 10
action set exten- 0 15
sion_queue=queue_$1@${domain_name}
action set fifo_simo=1 0 20
action set fifo_timeout=30 0 25
action set fifo_lag=10 0 30
action set fifo_destroy_after_use=true 0 35
action set fifo_extension_member=$1@${domain_name} 0 40
action lua extension_queue.lua 0 45
5.1. Dialplans 67
Wake Up
tion_number
action answer 0 10
action set time_zone_offset=- 0 20
7
action lua wakeup.lua 0 25
dx

condition destina- ^dx$ 0 5
tion_number
action answer 0 10
action read 11 11 0 15
‘tone_stream://%(10000,0,350,440)’
digits 5000 #
action transfer -bleg ${digits} 0 20
ATT Xfer

condition destina- âtt_xfer$ 0 5
tion_number
action read 2 6 0 10
‘tone_stream://%(10000,0,350,440)’
digits 30000 #
action set origination_cancel_key=# 0 15
action att_xfer user/${digits}@${domain_name} 0 20
Evesdrop
tion_number
action answer 0 10
action lua eavesdrop.lua 0 20
$1
Please Hold

condition ${user_exists} ^true$ 0 5
action set trans- 0 10
fer_ringback=$${hold_music}
action answer 0 15
action playback ivr/ivr- 0 25
hold_connect_call.wav
Cluecon Weekly

condition destina- ^*9(888|8888|1616|3232)$ 0 5
tion_number
action export hold_music=silence 0 10
action bridge sofia/${use_profile}/$1@conference.freeswitch.org 0 15
5.1. Dialplans 69
Bind Digit Action
Di- Di- Dialplan Detail Data Di- Di- Di- Di-

alplan alplan alplan alplan alplan alplan
condi- ${sip_authorized}
TRUE never 0 5
tion
action set bind_target=both TRUE 0 10
anti- set bind_target=peer TRUE 0 15
action
condi- 1 25
tion
action bind_digit_action
local,*1,exec:execute_extension,dx XML ${con- 1 30
text},${bind_target}
local,*2,exec:record_session,$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${s
1 35
local,*3,exec:execute_extension,cf XML ${con- 1 40
text},${bind_target}
local,*4,exec:execute_extension,att_xfer XML 1 45
${context},${bind_target}
action digit_action_set_realm
local 1 50
cf

condition destina- ^cf$ 0 5
tion_number
action answer 0 10
action transfer -both 30${di- 0 15
aled_extension:2}
XML ${context}
5.1.4 Dialplan Application
Dialplan Application uses FreeSWITCH show application to build the dropdown lists that are found in FusionPBX
dialplans. This is a list from a default install and the list can change depending on how many FreeSWITCH modules
are installed.
name description syntax

answer Answer the call
att_xfer Attended Transfer <channel_url>
bgsystem Execute a system command in the background <command>
bind_digit_action bind a key sequence or regex to an action <realm>,<digits|~reg
bind_meta_app Bind a key to an application <key> [a|b|ab] [a|b|o|
block_dtmf Block DTMF
break Break

bridge Bridge Audio <channel_url>
bridge_export Export a channel variable across a bridge <varname>=<value>
callcenter CallCenter queue_name
capture capture data into a var <varname>|<data>|<
check_acl Check an ip against an ACL list <ip> <acl | cidr> [<h
clear_digit_action clear all digit bindings <realm>|all[,target]
clear_speech_cache Clear Speech Handle Cache
cng_plc Do PLC on CNG frames
conference conference
conference_set_auto_outcall conference_set_auto_outcall
db Insert to the db [insert|delete]/<realm
decode_video decode picture [max_pictures]
deduplicate_dtmf Prevent duplicate inband + 2833 dtmf [only_rtp]
deflect Send call deflect <deflect_data>
delay_echo echo audio at a specified delay <delay ms>
detect_audio detect_audio <threshold> <audio_
detect_silence detect_silence <threshold> <silence
detect_speech Detect speech <mod_name> <gram
digit_action_set_realm change binding realm <realm>[,<target>]

displace_session Displace File <path> [<flags>] [+ti
early_hangup Enable early hangup
eavesdrop eavesdrop on a uuid [all | <uuid>]
echo Echo
enable_heartbeat Enable Media Heartbeat [0|<seconds>]
enable_keepalive Enable Keepalive [0|<seconds>]
endless_playback Playback File Endlessly <path>
enum Perform an ENUM lookup [reload | <number> [
eval Do Nothing
event Fire an event
execute_extension Execute an extension <extension> <dialpla
export Export a channel variable across a bridge <varname>=<value>
fax_detect Detect faxes
fifo Park with FIFO <fifo name>[!<impor
fifo_track_call Count a call as a fifo call in the manual_calls queue <fifo_outbound_uuid
fire fire the message
flush_dtmf flush any queued dtmf
gentones Generate Tones <tgml_script>[|<loop
group Manage a group [insert|delete]:<group
hangup Hangup the call [<cause>]
hash Insert into the hashtable [insert|insert_ifempty
hold Send a hold message [<display message>]
info Display Call Info
info Display Call Info
intercept intercept [-bleg] <uuid>
ivr Run an ivr menu
jitterbuffer Send session jitterbuffer <jitterbuffer_data>
limit Limit <backend> <realm>
limit_execute Limit <backend> <realm>
5.1. Dialplans 71

limit_hash Limit <realm> <id> [<max
limit_hash_execute Limit <realm> <id> <max>
log Logs to the logger <log_level> <log_str
loop_playback Playback File looply [+loops] <path>
media_reset Reset all bypass/proxy media flags
mkdir Create a directory <path>
multiset Set many channel variables [^^<delim>]<varnam
multiunset Unset many channel variables [^^<delim>]<varnam
mutex block on a call flow only allowing one at a time <keyname>[ on|off]
novideo Refuse Inbound Video
park Park
park_state Park State
phrase Say a Phrase <macro_name>,<dat
pickup Pickup [<key>]
play_and_detect_speech Play and do speech recognition <file> detect:<engine
play_and_get_digits Play and get Digits <min> <max> <tries
play_fsv play a fsv file <file>
play_yuv play a yvv file <file> [width] [heigh
playback Playback File <path>
pre_answer Pre-Answer the call
preprocess pre-process
presence Send Presence <rpid> <status> [<id
privacy Set privacy on calls off|on|name|full|num
push Set a channel variable <varname>=<value>
queue_dtmf Queue dtmf to be sent <dtmf_data>
read Read Digits <min> <max> <file>
record Record File <path> [<time_limit_
record_fsv record an fsv file <file>
record_session Record Session <path> [+<timeout>]
record_session_mask Mask audio in recording <path>
record_session_unmask Resume recording <path>
recovery_refresh Send call recovery_refresh
redirect Send session redirect <redirect_data>
remove_bugs Remove media bugs [<function>]
rename Rename file <from_path> <to_pa
reply reply to a message
respond Send session respond <respond_data>
ring_ready Indicate Ring_Ready
rxfax FAX Receive Application <filename>
say say <module_name>[:<la
sched_broadcast Schedule a broadcast in the future [+]<time> <path> [a
sched_cancel cancel scheduled tasks [group]
sched_hangup Schedule a hangup in the future [+]<time> [<cause>]
sched_heartbeat Enable Scheduled Heartbeat [0|<seconds>]
sched_transfer Schedule a transfer in the future [+]<time> <extensio
send send the message as-is
send_display Send session a new display <text>
send_dtmf Send dtmf to be sent <dtmf_data>
send_info Send info <info>

session_loglevel session_loglevel <level>
set set a variable
set Set a channel variable <varname>=<value>
set_audio_level set volume
set_global Set a global variable <varname>=<value>
set_media_stats Set Media Stats
set_mute set mute
set_name Name the channel <name>
set_profile_var Set a caller profile variable <varname>=<value>
set_user Set a User <user>@<domain> [
set_zombie_exec Enable Zombie Execution
sleep Pause a channel <pausemilliseconds>
socket Connect to a socket <ip>[:<port>]
sofia_sla private sofia sla function <uuid>
soft_hold Put a bridged channel on hold <unhold key> [<moh
sound_test Analyze Audio
spandsp_detect_tdd Detect TDD data
spandsp_inject_tdd Send TDD data
spandsp_send_tdd Send TDD data
spandsp_start_dtmf Detect dtmf
spandsp_start_fax_detect start fax detect <app>[ <arg>][ <tim
spandsp_start_tone_detect Start background tone detection with cadence <name>
spandsp_stop_detect_tdd stop sending tdd
spandsp_stop_dtmf stop inband dtmf
spandsp_stop_fax_detect stop fax detect
spandsp_stop_inject_tdd stop sending tdd
spandsp_stop_tone_detect Stop background tone detection with cadence
speak Speak text <engine>|<voice>|<t
start_dtmf Detect dtmf
start_dtmf_generate Generate dtmf
stop stop execution
stop Do Nothing
stop_displace_session Stop Displace File <path>
stop_dtmf stop inband dtmf
stop_dtmf_generate stop inband dtmf generation [write]
stop_record_session Stop Record Session <path>
stop_tone_detect stop detecting tones
stop_video_write_overlay Stop video write overlay <path>
stopfax Stop FAX Application
strftime strftime [<epoch>|]<format s
system execute a system command
system Execute a system command <command>
t38_gateway Convert to T38 Gateway if tones are heard
three_way three way call with a uuid <uuid>
tone_detect Detect tones
transfer Transfer a channel <exten> [<dialplan>
transfer_vars Transfer variables <~variable_prefix|va
txfax FAX Transmit Application <filename>
unbind_meta_app Unbind a key from an application [<key>]
5.1. Dialplans 73

unblock_dtmf Stop blocking DTMF
unhold Send a un-hold message
unloop Tell loopback to unfold
unset unset a variable
unset Unset a channel variable <varname>
unshift Set a channel variable <varname>=<value>
valet_park valet_park <lotname> <extensio
verbose_events Make ALL Events verbose.
video_decode Set video decode. [[on|wait]|off]
video_refresh Send video refresh. [manual|auto]
video_write_overlay Video write overlay <path> [<pos>] [<alp
wait_for_answer Wait for call to be answered
wait_for_silence wait_for_silence <silence_thresh> <si
5.1.5 Inbound Routes
Route incoming calls to destinations based on one or more conditions. It can send incoming calls to:
• IVR Menu
• Call Group
• Extension
• External Number
• Script
Directs public inbound calls to an internal destination on the system. Note that the only difference between the
inbound route dial plan and the normal dial plan is that the inbound route dial plan works on all calls that are in the
public context whereas the normal dial plan works on the domain context.
Inbound Call Routing is used to route incoming calls to destinations based on one or more conditions and context. It
can send incoming calls to an auto attendant, huntgroup, extension, external number, or a script. Order is important
when an anti-action is used or when there are multiple conditions that match.
Inbound routes can be used for advanced reasons. Dialplan > Destinations will create and configure the Inbound
Route for you.
• Name: The name of the Inbound Route.

• Number: The Number (DID) an outside caller will call.
• Context: Context of the Inbound Route. Usually will be public.

• Hostname: Usually blank, otherwise for advanced use.
• Order: Order where the inbound route will be used in the dialplan.
• Enabled: If the Inbound Route is enabled or disabled.
• Description: A way to organize what the inbound route is used for.
5.1.5.1 Edit/Add Inbound Routes
• Name: The name of the Inbound Route.

• Number: The Number (DID) an outside caller will call.
• Context: Context of the Inbound Route. Usually will be public.
• Order: Order where the inbound route will be used in the dialplan.
• Domain: Can be global to all domains or specific to one domain.
• Continue: If you want the call to continue through the order of the remaining dialplans. This is usually set as
false.
• Enabled: If the Inbound Route is enabled or disabled.
• Description: A way to organize what the inbound route is used for.
5.1.5.2 XML example
Route based on CallerID Name or Number.
5.1. Dialplans 75
Example used to send unwanted callers. (telemarketers that won’t stop)
<extension name="gotolennyCIDnumber" >

<condition field="context" expression="public"/>
<condition field="caller_id_number" expression="^1235554321$|^1235551234$">
<action application="answer"/>
<action application="bridge" data="sofia/${use_profile}/lenny@sip.itslenny.
˓→com:5060"/>
</condition>
</extension>
<extension name="gotolennyCIDname" >

<condition field="context" expression="public"/>
<condition field="caller_id_name" expression="^.*THE.*ANNOYING.*COMPANY.*$|^.*OTHER.
˓→*ANNOYING.*CALLER.*$">
<action application="answer"/>
<action application="bridge" data="sofia/${use_profile}/lenny@sip.itslenny.
˓→com:5060"/>
</condition>
</extension>
5.1.6 Outbound Routes
Route outbound calls to gateways, tdm, enum and more. When a call matches the conditions the call to outbound
routes. Check out the youtube video .
Configuring an Outbound Route.
• Select Dialplan from the drop-down list and then click Outbound Routes .
• Click the plus button on the right. Enter the route information below and Click Save once entry is complete.
Gateway: VoiceTel
Dialplan Expression: ^(?:\+?1)?(\d{10})$ (You can also choose more than one from the
˓→drop down list also as needed)
Order: 000
Enabled: true
5.1. Dialplans 77

Description: VoiceTel-out
By using VoiceTel you help support FusionPBX. Thank you for your support!
5.1.6.1 Pin Numbers
To have the system ask for a PIN number before a call is made. A good use is if you don’t want every user on the
system to be able to call international destinations. This can be done with a single PIN or multiple PINs by using the
“PIN Number APP”.
To use a single PIN number for all calls
Before the bridge action on the outbound route add the following actions
action set pin_number=(Whatever pin number you choose)

action lua pin_number.lua
To use the PIN Number App to manage multiple PINs
• First enable access to the “PIN Number” app by giving permissions to the group of users you want to have
access in Advanced > Group Manager. Make sure the “PIN Number” App is displayed in the menu by
selecting the groups that can view it in Advanced > Menu Manager.
• Set the PINs you would like to use in Apps > PIN Numbers
Before the bridge action on the outbound route add the following actions
action set pin_number=database

action lua pin_number.lua
Which gateway is my call using?
If you want to know the gateway your call is using there is currently no way to do this with FusionPBX’s GUI. Instead
you can do it this way.
• Go to Advanced -> Command and in the switch command dropdown section type
show channels as xml and then press the execute button.
• In the output that is returned, look for the string sofia/gateway/ and the gateway name. This is the gateway
your call is using.
5.1.7 Advanced Dialplans
FusionPBX installs several default dialplans. FusionPBX also gives the option to make new dialplans. This gives you
the power for more advanced functions, and produce the desired result.
5.1.7.1 Adding a Dialplan
You can create a new dialplan or copy and modify an existing dialplan.
• Go to Dialplan > Dialplan Manager
• Click the Plus icon at the top right.
• Complete required fields and click save.
5.1.7.2 Edit a Dialplan
Find the dialplan you want to edit and click the edit icon.
Once you enter data into the empty fields at the bottom and click save, more blank fileds will populate if needed.
5.1. Dialplans 79
5.1.7.3 Enable a Dialplan Destination
Dialplans that have a value in the Number filed can be enabled and used in Dialplan > Destinations. Setting the
destination field to True will enable the dialplan to be visable and used as an action in Dialplan > Destinations.
5.1.7.4 Dialplan example
This example will be for calling an extension on another tenant. This can be done several ways.
• We can use the adding a dialplan example and modify it for this example.
Cross Tenant Calling
This would require a prefix of 5 followed by 4 digit extensions. The prefix can be any number that you choose to use
and the 4 digit extension must match the destination tenant. So if the destination extensions are 3 digit then you would
use 3 instead of 4.
Tag Type Data Break In- Group Or-

line der
condi- ${destina- ^5(d{4})$ 5
tion tion_number}
action set domain_name=customer.domain.tld True 10
action set domain_uuid=correct-uuid-for-the- True 15
domain
action transfer $1 XML ${domain_name} 20
• Be sure to set the Continue dropdown box True

• Finally we have the desired dialplan to call from tenant A to tenant B.
5.1. Dialplans 81
Note: A quick way to find a domains uuid is by going to Advanced > Domains. Then click the edit icon on the
domain you want to know the uuid of. The uuid will be at the end of the url.
CHAPTER 6
Applications
6.1 Applications
In the Applications menu (Apps) section you will find Bridges, Call Block, Call Broadcast, Call Center, Call Detail
Records, Call Flows, Conference Center, Conference Controls, Conference Profiles, Contacts, Fax Server, Follow Me,
Grandstream Wave, IVR Menu, Music on Hold, Operator Panel, Phrases, Queues, Recordings, Ring Groups, Streams,
Time Conditions and Voicemail. Other apps can be added also.
6.1.1 Bridges
Bridge statements are used to send calls directly to other destinations like another PBX, Carrier or External SIP to
TDM Gateway and more. The bridge statements are added to destination select list.
• Click the Plus icon to add a bridge

• Click the edit icon on the right to edit a bridge
• Click the X to delete a bridge
83
6.1.1.1 Bridge Examples
Bridges are how ring groups are made. The code in FusionPBX simplifies that for you. You can however manually do
what ring groups do and with bridges.
Bridge Statement advanced options
• For multiple destinations. Multiple destinations are allowed as long as you use a , | or :_:
Comma , Means simultaneous

Pipe | Means In a sequence
colon under score colon :_: Means Enterprise
Loopback to an external number
loopback/12085551234
Loopback to multiple external numbers simultaneously
loopback/12085551234,loopback/12085552222,loopback/12085553333
To another sip server, sip gateway, or another carrier
sofia/internal/$1@xxx.xxx.xxx.xxx:5060
To a user
user/1001
or
sofia_contact(*/501@example.fusionpbx.com)
Using LCR
lcr/12085551234
Using variables
{abc=123}sofia/internal/$1@xxx.xxx.xxx.xxx:5060
Using variables in sequence with a sip server
{abc=123}sofia/internal/$1@xxx.xxx.xxx.xxx:5060|sofia/internal/$1@xxx.xxx.xxx.xxx:5060
Using variables in sequence with a sip server
[server=d1]sofia/internal/$1@xxx.xxx.xxx.xxx:5060|[server=d2]sofia/internal/$1@xxx.
˓→xxx.xxx.xxx:5060
6.1.2 Call block
A list of numbers from which to block calls.
84 Chapter 6. Applications
• To block a call click on the plus icon on the right

• Fill out the fields with pertinent information
Action:
Reject- Will reject the call
Busy- Will send a busy signal
Hold- Will put the call on hold
Voicemail- Will send the call to the specified voicemail box
Enhanced call-blocking introduced in Master branch 2.5.0: Call-blocking does an exact match on the inbound
caller-id number by default. This behaviour can be changed to use SQL “like” comparison or regex based comparison
by adding the following variable to the Default Settings:
6.1.3 Call Broadcast
Broadcast calls (a light dialer) to a defined list of phone numbers.
• To create a call broadcast click the plus on the right

Fill in the following fields
• Name- Name for the Call Broadcast.
6.1. Applications 85
• Accountcode Used by some billing systems.

• Timeout- Amount of time till hangup.
• Concurrent Limit- Used to pace the calls calls if the timeout was 60 and the concurent
limit is 100 then we would schedule 100 calls every 60 seconds.
• Caller ID Name- Name that will be used on outbound caller id.
• Caller ID Number- Number that will be used on outbound caller id.
• Destination Number- This is the internal number to call. Send the call to an IVR Menu or
some other number. If sending to a conference room make sure the room has a pin number or something that requires
user input you don’t want to add voiemail messages into the conference room.
For example *9198
• Phone Number List- List of phone numbers to call in the call broadcast.
This is the external number to call. Set a list of phone numbers one per row in the following format: 123-123-1234|Last
Name, First Name
5551231234|example 1
5551231234|example 2
5551231234|example 3
• Voicemail Detection- Set True or false to detect an answering machine.

• Description Help organize and label what the call broadcast is for.
• Once you have everything filled out click the Call Broadcast name you just created. On the top right click the
Send Broadcast button to start the call broadcast. To stop the call broadcast click STOP BROADCAST on the
top right.
Uses sched_api to schedule an API call in the future. Is used to schedule calls to the provided number/extensions and
send them to the extension an IVR Menu, Conference Room, or any other number. Could be used among other things
to schedule a Conference.
6.1.4 Call Center
List of queues for the call center.
6.1.4.1 Call Center Queues
• To add a Call Center Queue click the plus edit icon on the right
• Once a Queue is created click the edit pencil icon on the right. At the top right you can view, stop, start, restart
and save the queue
6.1.4.2 Call Center Agents
List of call center agents.
• From Apps > Call Center click Agents at the top right to access Call Center Agents
• Click the plus icon on the top right to add agents (make sure to set Agent ID!)
• Set the Agent Password, or add agent_authorized=true to the dialplan for *22 if you do not want to require a
PIN to log in
• If you want to enable Follow Me or Call Forwarding for an Agent, set the contact string to loopback/<extension>
6.1.4.3 Call Center Strategies
• Agent With Least Talk Time: Rings the Agent will ring that has the least time talking.
• Agent With Fewest Calls: Agent will ring that has the least calls.
• Longest Idle Agent: The agent will ring who idles the longest depending on their tier level.
• Ring All: All agents ring simultaneously.
• Random: Rings Agents will ring randomly in not particular order.
• Ring Progressively: Agents will ring the same as top-down and will progress until each agent ends up ringing.
• Round Robin: Will ring the next agent available in line.
• Sequentially By Agent Order: Agents will ring in a sequence by the tier and the tiers order.
• Top Down: Agent rings in order starting from one.
6.1.4.4 Agents
Select agents from the drop down list and specify tier level and tier position.
6.1.4.5 Music On Hold
Select the desired hold music. Music on hold, streams and ringtones can be used.
6.1.4.6 Record
Save the recording
6.1.4.7 Time base score
• Queue: Caller in queue time will start. If the caller goes to another queue the time will start over.
• System: Caller in queue will have their wait calculated as soon as they enter the system. If a caller chooses the
wrong queue, when they get to the correct queue the timer won’t start over again.
6.1.4.8 Max Wait Time
A value of 0 is the default and equals an infinate amount of time. Any other numeric value is calculated in seconds.
6.1.4.9 Max Wait Time with No Agent
Enter the max wait time with no agent. FusionPBX sets the default to 90 seconds and the Timeout Action will be
used if there are no agents available.
6.1.4.10 Max Wait Time with No Agent Time Reached
Enter the max wait time with no agent. FusionPBX sets the default to 30 seconds and the Timeout Action will be
used if there are no agents available.
6.1.4.11 Timeout Action
Set the action to perform when the max wait time is reached.
6.1.4.12 Tier Rules Apply
• True: Set the tier rule rules apply to true. The defined tiers will be used.
• False: Set the tier rule rules apply to false. All tiers will be used.
6.1.4.13 Tier Rule Wait Second
30 seconds is default. Enter the tier rule wait seconds.
6.1.4.14 Tier Rule Wait Multiply Level
• True: The amount of seconds the caller waits until the next tier. This value will increase(multiply) if Tier Rule
Wait Multiply Level is marked true.
• False: Tier Rule Wait Multiply Level is marked false then after the set amount of seconds pass the tiers in
order will execute with no wait.
6.1.4.15 Tier Rule No Agent No Wait
• True: Setting is enabled.

• False: Setting is disabled.
6.1.4.16 Discard Abandoned After
Default is 900 seconds. Sets the discard abandonded after seconds.
6.1.4.17 Abandoned Resume Allowed
• True: Setting is enabled. Permits a call to resume their posistion in the queue but only in the amount of seconds
set in discard abandonded after .
• False: Setting is disabled.
6.1.4.18 Caller ID Name Prefix
Set a prefix on the caller ID name.
6.1.4.19 Announce Sound
A sound to play to a caller every announce sound seconds. Needs the full path to the .wav file.
6.1.4.20 Announce Frequency
How often the announce sound is played in seconds.
6.1.4.21 Exit Key
Keys to quit the current queue waiting.
6.1.4.22 Description
Enter a description to help organize and define what the queue is for.
6.1.4.23 Agent Call Center Login
Agents can login to call center with *22 from the phone or via the FusionPBX web interface. Admin and Super Admin
accounts can also log other agents in or out.
• Login then go to Status > Agent Status
Call Center Default Settings
6.1.5 Call Detail Records
Call Detail Records (CDRs) are detailed information on the calls. Use the fields to filter the information for the specific
call records that are desired. Records in the call list can be saved locally using the Export button.
• CID Name Caller ID Name

• Source Where the call came from
• Destination Where the call went to
• Recording A link will appear if the call recorded
• Start Time the call entered the system
• TTA Time To Answer the call
• Duration How long the call was

• PDD Post Dial Delay
• MOS Mean Opinion Score is a measure of voice call quality
• Hangup Cause Details about the entire calls. Usually will be “Normal Clearing”
Call Detail Records are detailed information on the calls. The information contains source, destination, duration, and
other useful call details. Use the fields to filter the information for the specific call records that are desired. Then view
the calls in the list or download them as comma seperated file by using the CSV button.
Note that this page makes use of XML CDR for reporting.
6.1.5.1 Post Dial Delay (PDD)
Post Dial Delay (PDD) is experienced by the sender as the time from the sending of the final dialed digit to the point
at which the sender hears ring tone or other in-band information. In other words, the PDD would be the time from
when the sender sends the INVITE to receiving the first ringing response.
That said, PDD does not take into account the time it takes the receiver to hear the call coming in due to the various
factors on how they are setup for inbound calls. For example, call forwarding may affect the time it takes the receiver
to know that someone is calling because of call forwarding. The sender might hear a ring tone almost instantly from
the time it dials the final digit because they sent out an INVITE, but the receiver of the call might have setup inbound
calls to be forwarded to their cell phone, in which now the call must travel through their phone system, to their phone
system’s gateway carrier to deliver the sender’s call to the receiver’s cell phone carrier network in order for the cell
phone carrier to deliver the sender’s call to the receiver’s cell phone.
6.1.5.2 Recordings
Any calls which have the entry in the name column underlined (ie. the name is a link) have a recording available.
Clicking on the name will playback the recording in a new window. In such cases the number entry will also be a link
- clicking on this link will download the recording to your computer as a wav file.
6.1.5.3 Possible issues
No records showing up under Apps-Call Detail Records
Possible causes:
1. The module is disabled
• Make sure the XML CDR module is enabled and running in the Menu -> Advanced -> Modules.
2. Wrong xml_cdr.conf.xml config
• check <param name=”url” value=”http://127.0.0.1/app/xml_cdr/v_xml_cdr_import.php”/> and adapt it to your
situation.
• Compare your version (advanced-script editor-files-autoload_configs-xml_cdr.conf.xml) with the current de-
fault one that is included in FusionPBX (advanced-php editor-files-includes-templates-conf-autoload_configs-
xml_cdr.conf.xml). If it is different copy the default one over yours.
• Then edit the line <param name=”url” value=”http://{v_domain}/mod/xml_cdr/v_xml_cdr_import.php”/> and
replace {v_domain} with the domain or IP address of your FusionPBX server.
• Then edit the line <param name=”cred” value=”{v_user}:{v_pass}”/> and replace {v_user} with a complex
name of upper and lowercase and numeric characters so it is really ugly and secure, and do the same for v_pass.
• Make each of them completely unique.

• Be aware that these don’t have to match anything else on your server at all. This is because FusionPBX does
something very simple but clever here. The xml_cdr module uses this account when it does an http post to
FusionPBX of the new data. FusionPBX looks at the same xml_cdr.conf.xml file that the module uses in order
to check if the module is using a valid account and password. Since they both look at the same config file they
are using the same account and password and will happily talk to each other!
Once you’ve made these changes you can save the file. You could restart your server, or you could reloadxml and then
restart the xml_cdr module. Either is ok, it is up to you. Then your changes will have taken effect and you should no
longer lose your menu bar when looking at CDR information.
6.1.5.4 XML CDR configuration
For more detailed configuration go to the XML editor (Menu -> Advanced -> XML Editor) and in autoload configs
look at xml_cdr.conf.xml
Note: By default only the a-leg of the call is logged therefore if you make a recording of the b-leg you won’t be able
to retrieve it using the Call Detail Records. If you want the b-leg as well you need to change log-b-leg=true in this
config.
6.1.5.5 Harddrive space usage
Note: XML CDR data adds up fast, therefore you may need to clear this data at some point in
the future. By default freeswitch keeps this in (source install) /usr/local/freeswitch/log/cdr-csv or (pack-
age install) /var/log/freeswitch/xml_cdr and inside that by year, month and day. Recordings also take up
space and have to be manually deleted if you want the space back these are kept in (source install)
/usr/local/freeswitch/recordings/{Domian_Name} or (package install) /etc/freeswitch/recordings/{Domain Name}
and inside that by year, month and day.
6.1.5.6 CDR Default Settings
6.1.6 Call Flows
Direct calls between two destinations by calling a feature code.
• Name: Define the name of the call flow.

• Extension: Define what extension to use. (This will make an extension not already created)
• Feature Code: Define what * number to use

• Context: Domain context (typically leave as is)
• Status: Define what currently is in use.
• Pin Number: Define a pin number in order to execute either mode.
• Destination: Define where the call will go in the intial mode.
• Sound: Define the sound that will play once mode is engaged.
• Destination: Define what the destination will be.
• Alternative Label: Label that will show when alternative mode is in use.
• Alternative Sound: Define the sound that will play once alternative mode is engaged.
• Alternative Destination: Define where the call will go in the alternative mode.
• Description: Label what this call flow does.
6.1.6.1 Call Flow Example
In the Call Flow example below we have the name as Call Flow. Make the Extension number 30 that didn’t exist until
now. Create the feature code as a *code with *30. Keep the context as-is with training.fusionpbx.com . Select a Status
to show which mode. Make a PIN to help secure the call flow. Make the detination label as Day Mode. Select a sound
to auditorially indicate which mode is activated. Choose a destination for the alternative mode. Make the alternative
detination label as Night Mode. Select an alternative sound to auditorially indicate which mode is activated. Choose a
destination for the alternative mode. Finally, enter a description to describe what this call flow does.
6.1.7 Call Recordings
Shows the call recordings with name, length, date and time, and call direction.
• Click the eye icon on the right to view more details

• Click the X to delete a recording
• Click multiple check boxes to delete multiple at once.
6.1.8 Call Routing
Directs incoming calls for the extension
• To access call routing goto Accounts > click the edit pencil icon on the right of the extension
• Click CALL ROUTING on the top right
6.1.8.1 Call Forward and Do No Disturb
This will allow phones to sync CFWD and DND over SIP.
A few things need to be configured to enable this feature and restart freeswitch:
Uncomment this line in lua.conf.xml.
<hook event="PHONE_FEATURE_SUBSCRIBE" subclass="" script="app.lua feature_event"/>
Add to Default Settings:
Category = device
Subcategory = feature_sync
Type = boolean
Value = true
Enable Feature Sync on the Device
• Yealink
– Web Interface -> Features -> General Information -> Feature Key Synchronization set to Enabled
– Config Files -> features.feature_key_sync.enable
– Might be addition settings needed for the latest firmware. I tested with 81.0.110
• Polycom
– reg.{$row.line_number}.serverFeatureControl.cf=”1”
– reg.{$row.line_number}.serverFeatureControl.dnd=”1”
• Cisco SPA
– <Feature_Key_Sync_1_ group=”Ext_1/Call_Feature_Settings”>Yes</Feature_Key_Sync_1_>
6.1.9 Conference
Conferences is used to setup conference rooms with a name, description, and optional pin number.
Note: For advanced conferencing use Apps -> Conference Center
6.1.9.1 Conference Settings
• Name: Name for the conference.

• Extension: The number for the extension the user will dial.(Be sure it doesn’t exist before creating it.)
• Pin Number: If you want to add a layer of security to enter the conference.
• Profile:
– Default- The default audio quality rate and video.
– wait-mod- Wait Mod setting.
– wideband- Wideband audio quality rate and video.
– ultra-wideband- Ultra wideband quality rate and video.
– cdquality- CD Quality rate and video.
– page- Page setting.
• Flags: mute|deaf|waste|moderator (Other values are available also)
• Order: The order of the conference.
• Enabled: If the conference is enabled.
• Description: A way to organize what the conference purpose is.
6.1.9.2 Enable Conferences
By default Conferences use to be hidden from the menu.

• To add Conferences to the menu goto Advanced > Menu Manager and click the pencil edit icon on the right
• Then click the pencil edit icon on the right of Conferences
• Select from the Groups dropdown list superadmin and click add then save
6.1.10 Conference Center
Conference Centers are a group of conference rooms. They can be organized by cost center, geographically, or other
criteria.
• To Acess Conference Center goto Apps > Conference Center

• To view rooms click the ROOMS at the top right.
Note: For basic conferencing use Apps -> Conferences
6.1.10.1 Conference Center Options
• Name: Name of the Conference Center.

• Extension: Extension of the Conference Center. (Be sure to not use an extension already in use)
• Greeting: Choose a greeting to play.
• PIN Length: Add a layer of security for entering the Conference Center.
• Enabled: Enable or disable the Conference Center.
• Description: A way to organize what the Conference Center is for.
6.1.10.2 Conference Center Rooms
Apps > Conference Center > Click Rooms at the top right. This will take you to the Conference Center Rooms. From
here you can
• Create a Room
• Edit a Room

Conference Center Rooms Settings
6.1.11 Conference Controls
Call controls enable ability to assign digits to actions. They can be used to mute, unmute, or other actions during the
conference call.

• Click the edit icon on the right to adjust the control

• Click the plus to create a new control set
6.1.11.1 Default Conference Control
6.1.12 Conference Profiles
A group of conference parameters saved together as a profile.

• Click the edit icon on the right to adjust the profile

• Click the plus to create a new profile
6.1.12.1 Default Profile

• cdr-log-dir: Set as auto. Could be set manually and is enabled.

• domain: enabled.
• rate: The rate in kHz. 8000kHz and is enabled.
• interval: 20 is the default.
• energy-level: 15 is the default.
• auto-gain-level: 0 is the default.
• caller-controls: default is the default.
• moderator-controls: moderator is the default.
• muted-sound: conference/conf-muted.wav is the default.
• unmuted-sound: conference/conf-unmuted.wav is the default.
• alone-sound: conference/conf-alone.wav is the default.
• moh-sound: local_stream://default is the default.
• enter-sound: tone_stream://%(200,0,500,600,700) is the default.
• exit-sound: tone_stream://%(500,0,300,200,100,50,25) is the default.
• kicked-sound: conference/conf-kicked.wav is the default.
• locked-sound: conference/conf-locked.wav is the default.
• is-locked-sound: conference/conf-is-locked.wav is the default.
• is-unlocked-sound: conference/conf-is-unlocked.wav is the default.
• pin-sound: conference/conf-pin.wav is the default.
• bad-pin-sound: conference/conf-bad-pin.wav is the default.
• caller-id-name:
• caller-id-number:
• comfort-noise: true is the default.
6.1.13 Contacts
Contacts is a list of individuals and organizations.
• To create a contact click the plus and to edit a contact click the pencil icon on the right.
• Fill out the fields with pertinent information and click save.
• Users- Select the users that are allowed to view the contact
• Groups- Select the group that are allowed access to the contact.

• Go back into the contact to fill out more information that wasn’t available when you first created the contact.

• To generate a QR code click the QR CODE button at the top right

6.1.14 Fax Server
To receive a FAX setup a fax extension and then direct the incoming to it. Click here for the Youtube video
• New: Create a new fax to send.

• Inbox: Faxes received.
• Sent: Faxes sent.
• Log: Sucessful and failed attempts for both incoming and outgoing.
• Active: Shows the faxes in queue.
6.1.14.1 Fax Server Settings
There are more settings for fax under Advanced > Default Settings then fax category.
• To create a fax server goto App > Fax Server. Click the + on the right.

– Leave the Destination Number blank or faxing wont work.

• Destination Number is used in the Fax Server Dial Plan and is set based on the fax server internal extension
number.
• Define the fields, the ones in bold are required. It is a good idea to organize so define the name thoughtfully.
• The extension you must use one that is not allready created.
• Account Code should autofill. Again, leave the Destination Number blank.
• A prefix can be defined when sending a fax.
• Email is for inbound faxes and will be on the server and sent to the defines email.
• Define the Caller ID Name and Number.
• Leave the Forward Number and Greeting blank for normal settings.
• Number of channels define with a numerical value or keep blank for a default value.
• Keep organized by adding a Description.

6.1.14.2 New
To send a fax, the items in bold are required. To send a proper fax it is best to fill out all fields and attach any
documents. Keep in mind that the upload max MB is limited by Nginx and PHP config files.


6.1.14.3 Inbox
Click PDF to view the fax or right click on PDF and left click on Save Link As. If you defined and email address in
the email field you will receive the fax also to that email address.
6.1.14.4 Sent
Click PDF to view the fax or right click on PDF and left click on Save Link As.
6.1.14.5 Fax ATA
To connect to a fax machine with an ATA you will most likely need to adjust settings in the ATA web interface and in
FusionPBX.
Create an extension for the FAX machine. You can optionally set bypass media to true under advanced in the
extension settings.
6.1.14.6 FAX Default Settings
Menu -> Advanced -> Default Settings then category Fax

• Variables are used as defaults for the dialplan for sending and receiving faxes
• fax_enable_t38_request=false (Can be true or false)

• ignore_early_media=true (Can be true or false)
• Some carriers it’s better for fax_enable_t38_request=true and for some its better for it to be false.

• It’s best not to make an assumption and to do testing with different settings to get the best results for your
particular carrier.
• The variable fax_enable_t38_request=false will send a T38 reinvite when a fax tone is detected. In some cases
the re-invite always fails for some carriers which is why it is default to false.
Troubleshooting Tips
Faxing will fail at times. Fax Server should automatically try different methods for sending. There are different
combinations like;
• With T-38 on/off
• ECC on/off
• Sending a wav file
• Send a fax to HP faxback. This will test sending and receiving 1-888-473-2963
• Test sending with Faxtoy.net This will display what is faxed on their website. 1-855-330-1239 or 1-213-294-
2943
• Turn on verbose log in FreeSWITCH fax.conf.xml
– From your FusionPBX installation go to ADVANCED > XML Editor and a new window will open.
– Choose autoload_configs folder from the list, then choose fax.conf.xml.
– In fax.conf.xml there is an option that by default sets a variable called verbose = false. If you change
this to true you get more logging details as the fax is actually received, such as the quality of the
connection etc.
– You can see these details when you run the freeswitch command line ie. fs_cli
Command Line Fax Statistics
Grep from ssh or console access your freeswitch.log files for FAX_RETRY_STATS to start keeping track of suc-
cess/failure. Examples
Here’s how you can get some totals.
Total:
cat freeswitch.log |grep FAX_RETRY_STATS |wc -l
Success:
cat freeswitch.log |grep FAX_RETRY_STATS |grep SUCCESS |wc -l
Failures:
cat freeswitch.log |grep FAX_RETRY_STATS |grep FAIL |wc -l
6.1.15 Follow Me
Define alternate inbound call handling for the following extensions.

• Call Forward- (Disabled or Enabled) Input the destination number

• On Busy- (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
• No Answer- (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
• Not Registered- (Disabled or Enabled) If endpoint is not reachable, forward to this destination before going to
voicemail
• Follow Me- (Disabled or Enabled)
• Destinations- Can set Delay, Timeout and Prompt to accept the call.
• Ignore Busy- (Disabled or Enabled)
• Do Not Disturbe- (Disabled or Enabled)
This example has both the extension 1301 itself and and external number to call. If you don’t put the extension itself
the extension wont ring when in Follow Me. This is due to the flexible nature of FusionPBX where if you didn’t want
that extension to ring like if you were out of the office on a business trip.

6.1.15.1 Follow Me Default Settings
Click the link above for Follow Me default settings.
6.1.16 GS Wave
Grandstream Wave is a soft phone for smart phones or tablets. It can be configured easily with a QR code provided in
your FusionPBX installation.
• To use it download and install Grandstream Wave for your mobile device.
• Start the Grandstream Wave application on your mobile device.
• Then go to the Grandstream Wave Account Settings and press the plus+ to add a new account.
• Press on UCM Account (Scan QR Code) and then select the extension and scan the QR code.

You can choose any extension to provision the Grandstream Wave. Even If the extension is assigned to a desk phone.
Just be sure to enable multiple registrations.

Note: Be sure to assign a user to an extension for this application to be fully functional. This is a new app starting
with master branch version 4.5
6.1.17 IVR Menu
Welcome to the adding IVR section. Here you will find how to add and edit IVR’s.
• Click here for the youtube video
• Click on Apps then IVR Menu
• Click the Plus icon on the right

• Options in bold are mandatory.

• Name: Enter a name for the IVR menu
• Extension: Enter the extension number (This must a new extension that isn’t allready created)
• Greet Long: The long greeting when entering the menu.
• Greet Short: The short greeting is played when returning to the menu.
• Options: Define caller options for the IVR menu.
• Timeout: The number of milliseconds to wait after playing the greeting or the confirm macro.
• Exit Action: Select the exit action to be performed if the ivr exists.
• Direct Dial: Define whether the callers can dial directly to registered extensions.
• Ring Back: Defines what the caller will hear while the destination is being called.
• Caller ID Name Prefix: Set a prefix on the caller ID name.
• Enabled: set the status of the IVR Menu.

You can get very creative with IVR’s and are almost limitless in possibilities. In the basic example below we;
• Name the IVR “IVR Main”
• Extension “200”
• Greet Long a phrase that was made from the phrase section under apps
• Number entry in options, choose an extension for Destination and descriptions ie sales, billing, tech support,
and after hours. timeout 3000 milliseconds
• Exit Action to the extension 109 (after hours)
• Direct Dial to False and Ring back to Default.

You now have a list of IVR’s to go back to and edit or delete as needed.
6.1.17.1 IVR Default Settings
Click the link above for IVR default settings.
6.1.18 Operator Panel
Operator Panel is a simple and easy way to use the FusionPBX web interface to:
• Make calls from.
• See who is on a call.
• Eavesdrop on a call.

• Hangup your own call.

• Drag and drop blind transfer an active call.
• Drag and drop calling to other users.
• Login and out of queues and call center.
You can see the status of other users also depending on what permissions are set to the user.
Note: Make sure in Accounts > Extensions that the extension is assigned to the user. This will enable Operator Panel
for that user.
6.1.18.1 Operator Panel Status
• Available: The user will receive a call.

• On Break: The user won’t receive a call but can still receive a call from other users that directly call.
• Do Not Disturb: The user won’t receive any calls.
• Logged Out: The user won’t receive any calls as they are logged out.

6.1.19 Phrases
Create phrases of audio files to be played in sequence.
• Click the plus on the right to create a phrase and the pencil icon to edit a phrase
6.1.20 Music on Hold
Music on hold can be in WAV or MP3 format. To play an MP3 file you must have mod_shout enabled on the ‘Modules’
tab. You can adjust the volume of the MP3 audio from the ‘Settings’ tab. For best performance upload 16 bit,
8/16/32/48 kHz mono WAV files.

• Click the edit pencil on the right to customize music on hold options. This can be done on each kHz group.
• Name: Choose a name. (default is needed for the defauly Music on Hold.)
• Path: Path to where the music is.
• Shuffle: True or False (If true and multiple music files will shuffle the play order.)
• Sampling: The rate the music is encoded in.
• Channels: Mono or Stereo.
• Interval: Silence between files playing in milliseconds.
• Timer Name: Best to keep as soft.
• Chime File: The file you want to “chime in” while Music on Hold is playing.
• Chime Frequency: Seconds between each “chime in”.

• Chime Maximum: Max number attempts to “chime in”.

• Domain: Select Global for all domains or the specific domain for only that domain.
6.1.20.1 Music on Hold Tips
• When a new music on hold category mod_local_stream will be restarted. If it is busy then it will not restart
automatically. A manual restart of the module is required when it is not in use. The module can be restarted
from the Menu -> Advanced -> Modules or from the console and fs_cli with following command.
reload mod_local_stream
• Each music on hold category is given a name. If the domain is set to global the name will be the name in the
example below the protocol that is used is local_stream and the music on hold category is default and domain is
set to global.
local_stream://default
• It is possible that a domain or tenant can have its own category of music. In this example the name is ‘custom’
and the domain was assigned automatically to the current domain.
local_stream://domain_name/custom
6.1.21 Queues
Queues are used to setup waiting lines for callers. Also known as FIFO Queues.
The Queues feature is rarely used for call center type work. When needed, Call Center is usually used instead.

6.1.22 Recordings
Dial *732 to create a recording, or (for best results) upload a 16bit 8khz/16khz mono WAV file. Click here for the
youtube video.
To view and set the pin number goto Dialplan > Dialplan Manager > Click on Recordings > pin_number=8675309 at
the bottom.
Note: Pin number is recomended but can be left empty if no pin number is desired then pin_number=
6.1.22.1 Create a Recording
1. Dial *732 and wait for the voice prompt

2. Enter the password (pin_number) followed by the pound sign# Enter at least a 3 digit number. This will label
the recording file. (recording100.wav)
3. start talking to make the recording after the voice prompt and press the pound key #
4. Press 1 to accept the recording then hang up or press 2 to start over.

6.1.22.2 Edit Recording
1. Click the edit pencil icon.

2. Rename as needed.
3. Click save to save the changes.
6.1.22.3 Applying Recordings
Once you have a recording made you can use the recordings in different area’s of FusionPBX. Custom IVR’s and
phrases would be the typical uses.
6.1.22.4 Recordings Default Settings
Click the link above for Recordings default settings.
6.1.23 Ring Groups
A ring group is a set of destinations that can be called with a ring strategy.
To add a ring group click the plus. Click for the youtube video .

• Name A meaningful name for this ring group. This name is used in th Destination select list.
• Extension The extension number for this ring group.
• Greeting Play a sound file upon calling the Ring Group extension.
• Strategy The selectable way in which the destinations are being used.
– Simultaneous Rings all destinations. All destination share the same thread.
– Sequence Calls destinations in sequence where order that is lower goes first.
– Enterprise Ring all destinations. Each destination uses its own thread.
– Rollover Calls destinations in sequence and skips busy destinations.
– Random A random destination will ring.
• Destinations The destination numbers are the numbers for the ring group to call. Destinations can only be local
registered endpoints or external numbers.
– Extensions Local registered extensions.
– External numbers Destinations out to an external number.
• Prompt Where you determine if the call must have a dial to confirm before a pickup event.
• Caller ID Name Prefix The string that is added to the caller ID when it displays on the ringing extension.
• Caller ID Number Prefix The Number that is added to the caller ID when it displays on the ringing extension.
• Ring Back What the caller hears when they are waiting for the Destinations to answer. (ex. Music on Hold,
us-ring)
• Context The context defaults to the domain name.


6.1.23.1 Ring Group Example
In our example we will have 4 extensions all ring at the same time until one of them pick up first. Click the + to
create a ring group. Fill in the fields that are in bold. In the Extension box type a number that is NOT already created.
This new extention won’t be in the extension list. The strategy will be Simultaneous. Enter in the destination the 4
extensions 1001, 1002, 1003, 1004.
6.1.24 Streams
Define details for streaming audio.
• Make sure mod_shout is installed and is started.

• Have a shoutcast url ready to use. (shout://domain.tld/path/to/)
• To add a stream click the plus icon on the right
• Edit the fields:
– Name: Can be anything

– Location: Must start with shout://

– Enabled: If you want the stream enabled
– Domain: Choose a domain that will only have the stream. Choose Global for all domains
– Description: To help organize ;-)
Note: Editing a stream path will result in having to update anything that is using the stream. For example, if you have
extension 500 using stream “Local Weather” and you edit the shout:// path then you will have to go back to extension
500 and reset the music on hold for extension 500. This is by design.
Warning: Please be aware of your countries copyright laws for streaming the content you are going to stream.
6.1.25 Time Conditions
Dynamically route calls to an IVR menu, external numbers, scripts, or other destinations based on time conditions.
Fields in bold are mandatory.
• Name Name of the Time Condition.
• Extension Define an extension number that is NOT allready created.
• Presets US Holiday presets.
• Alternate Destination If the condition doesnt match the call will goto the defined alternate destination.
• Order Changes the order of which condition is evaluated first.
• Enabled If the ring group is enabled.

6.1.25.1 Time Conditions Example
In our example we have an employee that will receive calls during a set time range and set days. Below is what the
settings look like for Monday through Friday at 5:00pm to 11:00pm. If the employee doesnt answer the call will be
directed to the Timeout Destination. Label the Name as Oncall and invent the Extension as 10011. In the Settings
choose from the dropdown lists for Day of Week for the condition, Monday for the Value and Friday for the Range.

Next set of dropdown list choose Time of Day for the condition, 5:00 PM for the value and 11:00 PM for the Range.
If other options are needed just click the + to the right of Range.
The next dropdown choose the extension where the call is intended for. If the call is outside the date and time specified
the call will goto the Alternate Destination dropdown. Be sure Enabled is set True and click save.
6.1.25.2 Conditions
The most common conditions to use are Day of Week and Time of Day.
Time of Day
• Is a select list of every minute for the full 24 hour period of time.
Hour of Day
• Another alternative the Hour of Days. If you set a range of 9 - 4 it will include all of 4 until it changes to 5.
Day of Week
The day of week condition each day of the week is represented by a number. A valid range is from low to high. A
valid range is like Monday to Friday (2-6).
• 1 Sunday
• 2 Monday
• 3 Tuesday
• 4 Wednesday
• 5 Thursday
• 6 Friday
• 7 Saturday
An example of an invalid range would be Saturday to Sunday (7-1).
Time Conditions Default Settings
Click the link above for Time Conditions default settings.

6.1.26 Voicemail
To edit voicemail settings click the pencil edit icon on the right of the extension number.
Here you can edit voicemail settings.

• Play Tutorial- Play the voicemail tutorial after the next voicemail login
• Greeting- When you dial *97, record a greeting and set a number you can choose which greeting to use
• Alternate Greet ID- An alternative greet id used in the default greeting
• Options- Define caller options for the voicemail greeting
• Mail to- have voicemails emailed to this address
• Voicemail File- Select a listening option to include with the email notification
• Keep Local- Choose whether to keep the voicemail in the system after sending the email notification
• Forward Destinations- Forward voicemail messages to additional destinations
• Enabled- Enable or disable the voicemail box

Note: Starting version 4.2 remote access to voicemail by interrupting the greeting message by pressing “*” and
entering the password is disabled by default.
To enable remote access to voicemail

1. Go to your Fusionpbx installation menu.
2. Advanced.
3. Default Settings.
4. Voicemail category.
5. Enable and set true remote_access.
6.1.26.1 Voicemail Options
To access an extensions voicemail away from the extension.

• Dial the extension and interupt the greeting with the *star key.
*97 To access that extensions voicemail from the extension or the voicemail button
*98 To access any extensions voicemail
*99[ext] To access a specific extension voicemail

Main Menu
press 5 For advanced options
Advanced Options
press 1 Record a greeting
press 2 Choose a greeting
press 3 Record name
press 6 Change password
press 0 For main menu
Email Setup/Default Settings
Click the link above for setting up email server settings. These are the settings needed to enable your FusionPBX
installation to be able to send email notifications.
Voicemail Default Settings
Voicemail default settings gives the options to adjust voicemail settings on your FusionPBX installation globally.
Variables
These variables can be set in advanced -> variables or in the dialplan.
Name Value
skip_greeting true or false
skip_instructions true or false
voicemail_greeting_number 0-9
vm_disk_quota 0-3600 seconds
vm_message_ext wav or mp3
voicemail_authorized true or false
vm_say_caller_id_number true or false
vm_say_date_time true or false
Wav file is the default voicemail message file type. MP3 requires mod_shout to be installed and running.
Not Found Message
When an extension is unavailable and no voicemail is configured, there is an option to play a message to the caller
alerting them to this.
To enable/disable this, change the option for the not_found_message setting in Advanced > Default Settings >
Voicemail category to suit your preference.
Please note that enabling this option means that the call must be answered in order to play the message to the caller
and so the call will complete with a 200 OK rather than a 480 Unavailable or 486 Busy. In some jurisdictions this
could potentially be illegal as it turns an otherwise toll free call into a chargeable one.
6.1.26.2 Voicemail Transcription

FusionPBX supports Voicemail Transcription, where emails will include a transcribed version of the voicemail the
email was sent in regards to. To configure this feature, see applications/voicemail_transcription.rst.


CHAPTER 7
Status
7.1 Status
In the Status menu you have the options for Active Call Center, Active Calls, Active Conferences, Active Queues,
Agent Status, CDR Statistics, Emails, Extension Summary, Log Viewer, Registrations, Services, SIP Status, System
Status and Traffic Graph.
7.1.1 Active Call Center
Select a Call Center Queue from the list below to view its activity.
From here you can view status, evesdrop on the call, transfer the call or click to call an available agent.
137
Click to learn more about Call Center. Applications > Call Center
7.1.2 Active Calls
Use this to monitor and interact with the active calls.
Here you can view the sip profile used, time the call was created, number, cid number, destination, application, Codecs
used, and if the call is secure (encrypted)
• Click the X to end the call
• Click the Show All button to show calls in all domains.
7.1.3 Active Conferences
List all the conferences that are currently active with one or more members.
• Click view to view the active conference.
138 Chapter 7. Status

• Red ball icon: If illuminated the conference is being recorded.

• Lock: Can lock the conference from anyone else joining.
• Unmute All: Unmute all the members.
• End Conference: End the conference.
• CID Name: Caller ID Name
• CID Number: Caller ID Number
• Capabilities: Icons show what capabilities each member have like hear/mute, talking, and video.
• Joined: How long ago the member joined.
• Quiet: How long since the member was talking last.
• Has Floor: Who is currently talking.
• Mute: Mute a member.
• Dead: Make it so the member can’t hear what is being said in the conference.
• Kick: Kick the member from the conference.
7.1.4 Active Queues
Queues feature generates a dialplan that uses mod_fifo. FIFO stands for ‘first in first out’ in other words a queue.
7.1.5 Agent Status
List all the call center agents with the option to change the status of one or more agents.
7.1. Status 139

7.1.6 CDR Statistics
Call Detail Records Statics summarize the call information.
7.1.6.1 Definitions
• Hours: Specific hour in that day.

• Date: Specific date in that month.

• Time: Specific time in that day.
• Volume: Number of calls.
• Minutes: Specific number of minutes.
• Calls Per Minute: Specific number of calls per minute.
• Missed: Specific number of missed calls.
• ASR: The answer to seizure ratio. Which is how many calls where answered versus not answered.
• Aloc: ALOC is the average length of call.
• Days: Specific day in that month.
7.1.7 Emails
Manage failed email messages. If for some reason the message doesn’t get sent they will sit in a queue. You can view,
download or resend each message.
• Sent- Date and time last attempt to email was made

• Type- If the email was a missed call or voicemail
• Status- Status of the email
• Message- View, Download or Resend the email
• Reference- CDR information
• Eye icon- More details about the email
• X icon- Deletes the email
7.1.8 Extension Summary
Summary of extension activity per domain such as misssed calls, answered calls, no answer, inbound duration, out-
bound duration, number of outboud calls, number of inbound calls and Average length of Call (ALOC). The summa-
rized information can be downloaded as a CSV file.
7.1. Status 141

7.1.8.1 Definitions
• Extension: The extension number.

• Number Alias: Alias name for the extension number.
• Missed: Number of missed calls.
• No Answer: Number of calls not answered.
• Busy: Number of calss not answered wile busy.
• ALOC: The average length of call.
• Inbound Calls: Number of calls in.
• Inbound Duriation: Number of call minutes in.
• Outbound Calls: Number of calls out.
• Outbound Duriation: Number of call minutes out.
7.1.9 Log Viewer
View recent PBX activity and option to download the logs.

• Filter- Filter by specfic input

• Show Line Numbers- Shows the line numbers on the left side if checked
• Sort Decending- Sorts by decending
• Display- The ammount of log to display
• Reload Button- Reloads the log with filter, show line numbers, sort decending and display KB values
• Download- Downloads the log
7.1.10 Registrations
View the devices that are registered. This will show User, Agent, IP, Port Number, Hostname and Status. You can also
UNREGISTER, PROVISION and REBOOT supported devices from here.
7.1. Status 143

7.1.11 Services
Shows a list of processes, the status of the process and provides control to start and stop the process.
Click the plus on the right to add a service.

7.1.12 SIP Status
This will show sofia status of internal, internal-ipv6, external, and external-ipv6 profiles.
With profiles you can see
• REGISTRATIONS
• START/RESTART/RESCAN/FLUSH REGISTRATIONS
• You can also FLUSH CACHE
• RELOAD ACL
• RELOAD XML and REFRESH
• View UP time, sessions since startup, max sessions, and current stack size/max.
7.1.13 System Status
System Information, FusionPBX Version, Git Version, Switch Version, Memory Information, CPU Information, Hard
Drive Information and Memcache Information.
7.1. Status 145

7.1.14 Traffic Graph
Scalable Vector Graphics (SVG) support in your browser is required to view the traffic graph.

7.1. Status 147


CHAPTER 8
Advanced
8.1 Advanced
In the Advanced menu you will find Access Controls, App Manager, Backup, Command, Databases, Default Settings,
Domains, Grammer Editor, Group Manager, Menu Manager, Modules, Number Translations, PHP Editor, Provision
Editor, Sip Profiles, Script Editor, Settings, Transactions, Upgrade, Variables and XML Editor.
8.1.1 Access Controls
Access control list can allow or deny ranges of IP addresses. There are several purposes for using the ACL.
• The main purpose is for your carriers ip addresses. Add the carrier IP addresses to the CIDR.
• Be careful with what and how you use ACL.
• Most common mistakes result in calls not working between extensions and other undesirable results.
• Be sure to keep Domains access control to default deny.
• Do not put your public ip or phone IP addresses in the domains access control list.
• Don’t supply both the domain and the cidr on the same node.
• If adding a single IP address to the CIDR field make sure to add /32 on the end of the IP address.
8.1.1.1 Access Control Example
Goto Advanced > Access Controls. Click the edit icon for domains. At the bottom under nodes click the plus icon.
Type choose allow

CIDR enter the 12.34.56.0/32
Domain (Leave Blank, used for advanced scenarios)
Description (Carrier Name)
149
Click save
Goto > Status > Sip Status and click reloadacl.
Under Status > log viewer you should notice the ip added. This can be seen also from command line fs_cli by using
reloadacl
[NOTICE] switch_utils.c:545 Adding 12.34.56.0/32 (allow) [] to list domains
8.1.2 App Manager
For future use. Manage the applications that are installed.
8.1.3 Command
Provides a conventient way to execute system, PHP, switch and SQL commands.
150 Chapter 8. Advanced

• Click the drop down box on the right to choose from Switch, PHP, Shell and SQL to execute commands.
8.1.4 Databases
Database information. Most FusionPBX installs use Postgresql for FusionPBX and SQLite for the switch. This section
is for edge case installs.
8.1.5 Default Settings
Default Settings used for all domains. Branding can be done in this section, adjust or copy settings to specific domains
can be done in this section also.
8.1. Advanced 151

Default Settings have several different categories. Click on the category to view more details.
8.1.5.1 Adminer
FusionPBX menu Advanced > Adminer

FusionPBX version 4.2+ has Adminer disabled by default. To use Adminer, you must enable this option with True.
Default Setting Default Set- Default Set- Default Setting Default Setting Description
Subcategory ting Name ting Value Enabled
auto_login boolean TRUE FALSE This must be enabled in order
to use Adminer.
8.1.5.2 Cache
Option to use file cache for xml and not memcache.
Default Setting Sub- Default Setting Default Setting Default Setting Default Setting De-
category Name Value Enabled scription
method text memcache TRUE Cache methods file or
memcache.
location text /tmp TRUE Location for the file
cache.
8.1.5.3 Call Center
FusionPBX menu Apps > Call Center

Defaults for the amount of agent rows for Call Center.
agent_add_rows numeric 5 TRUE Number of default
“add” rows.
agent_edit_rows numeric 1 TRUE Number of default
“edit” rows.
8.1.5.4 CDR
FusionPBX menu Apps > CDR

CDR Stat hour limit, call leg, format, limit, http_enabled, archive database, and storage type settings can be set here.
Default Setting Default Setting Default Setting Default Setting Default Setting Descrip-
Subcategory Name Value Enabled tion
stat_hours_limit numeric 24 FALSE
b_leg array outbound FALSE
b_leg array inbound FALSE
b_leg array local FALSE
format text json TRUE
limit numeric 800 TRUE
http_enabled boolean TRUE TRUE
archive_database_driver text pgsql FALSE Archive Database Driver
archive_database_host text FALSE IP/Hostname of Archive
Database
archive_database_password
text FALSE Archive Database Password
archive_database_port text 5432 FALSE Archive Database Port
archive_database_username
text FALSE Archive Database Username
storage text db TRUE
archive_database boolean FALSE FALSE Enable Dedicated CDR
Database Access
archive_database_name text fusionpbx FALSE Archive Database Name
8.1.5.5 Dashboard
FusionPBX menu Home > Dashboard

Different user level settings that control what is seen and not seen on the dashboard for each user access level.
8.1. Advanced 153

Default Setting Default Default Default Set- Default Setting Description

Subcategory Setting Setting ting Enabled
Name Value
admin array voicemail TRUE Enable Dashboard Voicemail block for users
in the admin group.
admin array missed TRUE Enable Dashboard Missed Calls block for
users in the admin group.
admin array recent TRUE Enable Dashboard Recent Calls block for
admin array limits FALSE Enable Dashboard Domain Limits block for
admin array counts TRUE Enable Dashboard Domain Counts block for
admin array ring_groups TRUE Enable Dashboard Ring Group Forwarding
controls for users in the admin group.
admin array caller_id FALSE Enable changing Caller ID name and number.
superadmin array voicemail TRUE Enable Dashboard Voicemail block for users
in the superadmin group.
superadmin array missed TRUE Enable Dashboard Missed Calls block for
users in the superadmin group.
superadmin array recent TRUE Enable Dashboard Recent Calls block for
superadmin array limits FALSE Enable Dashboard Domain Limits block for
superadmin array counts TRUE Enable Dashboard System Counts block for
superadmin array call_routing TRUE Enable Dashboard Call Routing controls for
superadmin array caller_id FALSE Enable changing Caller ID name and number.
superadmin array ring_groups TRUE Enable Dashboard Ring Group Forwarding
controls for users in the superadmin group.
user array voicemail TRUE Enable Dashboard Voicemail block for users
in the users group.
user array missed TRUE Enable Dashboard Missed Calls block for
users in the users group.
user array recent TRUE Enable Dashboard Recent Calls block for
user array call_routing TRUE Enable Dashboard Call Routing controls for
user array ring_groups TRUE Enable Dashboard Ring Group Forwarding
controls for users in the users group.
user array caller_id FALSE Enable changing Caller ID name and number.
admin array call_routing TRUE Enable Dashboard Call Routing controls for
superadmin array system TRUE Enable Dashboard System Status block for
agent array call_center_agents
TRUE Enable Dashboard Call Center Agent Status
block for users in the agent group.

8.1.5.6 Destinations
FusionPBX menu Dialplan > Destinations

Destinations specific defaults.
dialplan_details boolean TRUE TRUE
8.1.5.7 Domains
FusionPBX menu Advanced > Domains

Domain specific defaults.
Default De- Default Setting Value Default Default Setting De-

Setting fault Setting scription
Subcat- Set- En-
egory ting abled
Name
dial_string text {sip_invite_domain=${domain_name},leg_timeout=${call_timeout},presence_id=${dialed_user}@${dialed_d
TRUE The dial string used
template name default TRUE The template used
menu uuid b4750c3f-2a86-b00d-b7d0-345c14eca286 TRUE The menu uuid
language code en-us TRUE Choose the language
cidr array FALSE Allow only specific ip
addresses access
country code us TRUE The country code
bridge text outbound TRUE out-
bound,loopback,lcr
paging nu- 100 TRUE Set the maximum
meric number of records
displayed per page.
(Default: 50)
time_zone name America/Los_Angeles TRUE Time zone used. Fol-
lows UNIX format
8.1.5.8 Editor
FusionPBX menu Advanced > php editor, grammar editor, provision editor, and xml editor.
Editor specific defaults.
8.1. Advanced 155

Default Setting Default Set- Default Set- Default Set- Default Setting Description
Subcategory ting Name ting Value ting Enabled
indent_guides boolean FALSE FALSE Set the default visibility of indent guides
for Editor.
invisibles boolean FALSE FALSE Set the default state of invisible characters
for Editor.
line_numbers boolean FALSE FALSE Set the default visibility of line numbers
for Editor.
theme text Cobalt FALSE Set the default theme.
font_size text 14px FALSE Set the default text size for Editor.
live_previews boolean FALSE FALSE Enable or disable live previewing of syn-
tax, text size and theme changes.
8.1.5.9 Email
This is where you configure email settings to receive email notifications of voicemail, missed calls and fax.
Here are some example settings for some of the most common email providers.
• SMTP2GO
• GMAIL
Default Setting Default Default Setting Default Set- Default Setting Description
Subcategory Setting Value ting Enabled
Name
smtp_host text mail.server.provider.com
TRUE email providers server address
smtp_from text emailexam- TRUE smtp from emaill address
ple@emailprovider.com
smtp_port numeric 587 TRUE port number of the mail server provider
smtp_from_name text Voicemail TRUE smtp from name
smtp_auth text TRUE TRUE If smtp auth is required
smtp_username text user name TRUE typically the email user name
smtp_password text supersecurepass- TRUE typically the email password
word!
smtp_secure text tls TRUE tls or ssl depending on the provider.
smtp_validate_certificate
boolean TRUE TRUE set to false to ignore SSL certificate
warnings e.g. for self-signed certificates
method text smtp TRUE smtp|sendmail|mail|qmail
Error log for failed or sucessfully sent messages.

• Email Log
8.1.5.10 Fax
Apps > Fax Server

Specific default settings for fax server.

Default Set- Default Default Set- Default Default Setting Description

ting Subcate- Setting ting Value Setting
gory Name Enabled
cover_logo text TRUE Path to image/logo file displayed in the header
of the cover sheet.
al- array .pdf TRUE Allowed extension to send .pdf
lowed_extension
al- array .tif TRUE Allowed extension to send .tif
lowed_extension
al- array .tiff TRUE Allowed extension to send .tiff
lowed_extension
cover_header text FALSE Default information displayed beneath the
logo in the header of the cover sheet.
page_size text letter TRUE Set the default page size of new faxes.
resolution text fine TRUE Set the default transmission quality of new
faxes.
variable array fax_enable_t38=true
TRUE Enable T.38.
variable array fax_enable_t38_request=false
TRUE Send a T38 reinvite when a fax tone is de-
tected.
variable array ig- TRUE Ignore ringing to improve fax success rate.
nore_early_media=true
keep_local boolean TRUE TRUE Keep the file after sending or receiving the fax.
send_mode text queue FALSE Send mode. queue is default.
send_retry_limit numeric 5 TRUE Number of attempts to send fax (count only
calls with answer).
send_retry_intervalnumeric 15 TRUE Delay before we make next call after answered
call.
send_no_answer_retry_limit
numeric 3 TRUE Number of unanswered attempts in sequence.
send_no_answer_retry_interval
numeric 30 TRUE Delay before we make next call after no an-
swered call.
send_no_answer_limit
numeric 3 TRUE Giveup reach the destination after this number
of sequences.
send_no_answer_interval
numeric 300 TRUE Delay before next call sequence.
storage_type text base64 FALSE Store FAX in base64.
smtp_from text TRUE SMTP from address.
smtp_from_name text TRUE SMTP from name. Depends on the server, can
be full email or everything before the @ sign.
cover_font text times FALSE Font used to generate cover page. Can be full
path to .ttf file or font name alredy installed.
cover_footer text TRUE Notice displayed in the footer of the cover
sheet.
8.1.5.11 Follow Me
FusionPBX menu Apps > Follow Me

Specific defaults for Follow Me.
8.1. Advanced 157

max_destinations numeric 5 FALSE Set the maximum number of Fol-
low Me Destinations.
timeout numeric 30 FALSE Set the default Follow Me Timeout
value.
8.1.5.12 Ivr Menu
FusionPBX menu Apps > IVR Menus

Specific default for IVR Menu.
option_add_rows numeric 5 TRUE Number of default
“add” rows.
option_edit_rows numeric 1 TRUE Number of default
“edit” rows.
8.1.5.13 Limit
Limit specific default settings.
call_center_queues numeric 3 FALSE Limit used in Call Center
Queues.
destinations numeric 3 FALSE Limit used in Destina-
tions.
devices numeric 3 FALSE Limit used in Devices.
extensions numeric 3 FALSE Limit used in Extensions.
gateways numeric 3 FALSE Limit used in Gateways.
ivr_menus numeric 3 FALSE Limit used in IVR
Menus.
ring_groups numeric 3 FALSE Limit used in Ring
Groups.
users numeric 3 FALSE Limit used in Users.
8.1.5.14 Login
Login specific default settings.

Default Set- Default Default Set- Default Default Setting Description

ting Subcate- Setting ting Value Setting
gory Name Enabled
pass- text 9pG6sgerhuh5hetjnsrtjrjrdW
FALSE Display a Reset Password link on the login
word_reset_key box (requires smtp_host be defined).
do- boolean TRUE FALSE Displays a domain input or select box (if do-
main_name_visible main_name array defined) on the login box.
domain_name array pbx1.yourdomain.com
FALSE Domain select option displayed on the login
box.
message text Welcome to TRUE Display a message at login.
FusionPBX!
8.1.5.15 Provision
In the Provisioning section, there are a few key options that have to be set in order to turn auto provisioning on.
• enabled: Must be enabled and set to value true and enabled True. It is disabled by default.
• http_auth_username: Must be enabled and set to value true and enabled True. It is disabled by default. Be
sure to use a strong username.
• http_auth_password: Must be enabled and set to value true and enabled True. It is disabled by default. Be
sure to use a strong password.
Default Setting Subcategory Default Setting Name Default Setting Value

fanvil_time_zone text -20
fanvil_time_zone_name text UTC-5
fanvil_location numeric 4
fanvil_realm text enter a value
fanvil_greeting text FusionPBX
fanvil_date_display numeric 3
fanvil_time_display numeric 1
fanvil_wifi_enable numeric 0
fanvil_stun_port numeric 3478
grandstream_call_waiting text 0
contact_grandstream boolean TRUE
grandstream_gxp_time_zone text auto
grandstream_check_sip_user_id text 1
grandstream_config_server_path text none
grandstream_firmware_path text mydomain.com/app/provision
grandstream_lan_port_vlan text 1
grandstream_pc_port_vlan text 1
grandstream_ldap_base_dn text dc=mydomain,dc=com
grandstream_ldap_display_name text givenName sn title
grandstream_ldap_mail_attr text mail
grandstream_ldap_mail_filter text (mail=%)
grandstream_ldap_name_attr text givenName sn title mail
grandstream_ldap_name_filter text (cn=%)
grandstream_ldap_number_attr text telephoneNumber mobile homePhone
grandstream_ldap_number_filter text (|(telephoneNumber=%)(homePhone=%)(moblie=%))
grandstream_ldap_password text super-secret
8.1. Advanced 159


grandstream_ldap_server text mydomain.com
grandstream_ldap_user_base text ou=users,dc=mydomain,dc=com
grandstream_ldap_username text cn=pbxadmin,dc=mydomain,dc=com
grandstream_phonebook_download_interval text 720
grandstream_qos_rtp text 5
grandstream_qos_sip text 3
grandstream_sip_only_known_servers text 1
grandstream_stun_server text mydomain.com
grandstream_validate_incoming_sip text 1
grandstream_wallpaper_url text https://mydomain.com/files/wallpaper.jpg
grandstream_bluetooth_power text 1
grandstream_bluetooth_handsfree text 1
grandstream_auto_attended_transfer text 1
grandstream_syslog_server text
grandstream_syslog_level text 0
grandstream_send_sip_log text 0
grandstream_screensaver text 1
grandstream_screensaver_source text 0
grandstream_screensaver_show_date_time text 1
grandstream_screensaver_timeout text 5
grandstream_screensaver_server_path text
grandstream_screensaver_xml_download_interval text 0
grandstream_srtp text 0
htek_time_zone text 18
htek_dst numeric 1
htek_date_display_format numeric 1
htek_time_format numeric 1
polycom_digitmap text [*]xxxx|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-
polycom_call_waiting text 1
cidr array 209.210.17.193/32
http_auth_username text admin
http_auth_type text digest
enabled text TRUE
cidr array 209.210.16.196/32
auto_insert_enabled boolean TRUE
http_auth_disable boolean FALSE
admin_name text
admin_password text
path text
outbound_proxy_primary text
outbound_proxy_secondary text
line_sip_port numeric 5060
line_sip_transport text tcp
daylight_savings_enabled boolean TRUE
daylight_savings_start_month text 3
daylight_savings_start_weekday text 7
daylight_savings_start_time text 2
daylight_savings_stop_weekday text 7
daylight_savings_stop_time text 2


http_domain_filter boolean TRUE
contact_users boolean TRUE
contact_groups boolean TRUE
number_as_presence_id text TRUE
ntp_server_primary text pool.ntp.org
ntp_server_secondary text 2.us.pool.ntp.org
spa_time_zone text GMT-07:00
spa_time_format text 12hr
spa_date_format text day/month
spa_back_light_timer text 30 s
spa_handle_via_rport text Yes
spa_insert_via_rport text Yes
spa_call_waiting text Yes
spa_feature_key_sync text No
spa_dual_registration text No
spa_register_when_failover text No
snom_call_waiting text on
nway_conference text TRUE
vtech_vlan_wan_enable text 0
vtech_vlan_wan_id text 1
vtech_vlan_wan_priority text 0
stun_server text
stun_port numeric 3478
aastra_gmt_offset numeric 0
aastra_time_format numeric 0
aastra_date_format numeric 0
yealink_time_zone text -5
yealink_time_zone_name text United States-Eastern Time
yealink_time_format text 1
yealink_rport boolean 1
yealink_session_timer boolean 0
yealink_retransmission boolean 0
yealink_subscribe_mwi_to_vm boolean 1
yealink_srtp_encryption text 0
yealink_rfc2543_hold numeric 0
yealink_blf_led_mode numeric 0
yealink_trust_ctrl numeric 1
yealink_direct_ip_call_enable numeric 0
yealink_hide_feature_access_codes_enable numeric 0
yealink_voice_mail_popup_enable numeric 0
yealink_missed_call_popup_enable numeric 0
yealink_cid_source numeric 0
yealink_dtmf_hide numeric 1
yealink_sip_listen_port numeric 5060
yealink_firmware_url text https://server.yourdomain.com/app/yealink/resources/fi
yealink_firmware_cp860 text cp860-37.81.0.10.rom
yealink_firmware_cp960 text cp960-73.80.0.25.rom
yealink_firmware_t29g text t29g-46.81.0.110.rom
8.1. Advanced 161


yealink_firmware_t40p text t40p-54.81.0.110.rom
yealink_firmware_t41s text t41s-66.81.0.110.rom
yealink_firmware_t54s text T54S(T52S)-70.82.0.20.rom
yealink_firmware_t56a text t56a-58.80.0.25.rom
yealink_firmware_t58a text t58a-58.80.0.25.rom
yealink_firmware_t58v text t58v-58.80.0.25.rom
yealink_firmware_vp530 text vp530-23.70.0.40.rom
yealink_network_vpn_enable boolean 1
yealink_ip_address_mode numeric 0
yealink_lldp_enable boolean 0
yealink_cdp_enable boolean 0
yealink_overwrite_mode boolean 0
yealink_dsskey_length numeric 0
yealink_feature_key_sync numeric 0
yealink_predial_autodial boolean 0
yealink_ring_type text custom.wav
yealink_ringtone_delete text http://localhost/all,delete
daylight_savings_start_day text 11
daylight_savings_stop_month text 11
daylight_savings_stop_day text 4
http_auth_password array 555
fanvil_stun_server text example.domain.tld
grandstream_dns_mode text 1
grandstream_global_contact_groups text contacts_elementary,contacts_facilities,contacts_other
grandstream_nat_traversal text 0
grandstream_phonebook_xml_server_path text mydomain.com/app/provision/pb/
polycom_gmt_offset text
polycom_feature_key_sync numeric 0
voicemail_number text *97
line_register_expires numeric 120
contact_extensions boolean TRUE
spa_dial_plan text (*xxxxxxx|*xxxxxx|*xxxxx|*xxxx|*xxx|*xx*|*x|**xx
spa_secure_call_setting text No
snom_time_zone text USA-7
yealink_date_format text 3
yealink_outbound_proxy_fallback_interval numeric 3600
yealink_missed_call_power_led_flash_enable numeric 0
yealink_firmware_t41p text t41p-36.81.0.110.rom
yealink_openvpn_url text hxxps://replace-this.url/openvpn.tar
yealink_ringtone_url text custom.wav


yealink_call_waiting text 0
grandstream_dial_plan text {x+|*x+|*++|park+*x+|flow+*x+}
8.1.5.16 Recordings
FusionPBX menu Apps > Recordings

Recordings specific default settings.
storage_type text base64 FALSE Save recordings in the database
in base64 format.
8.1.5.17 Ring Group
FusionPBX menu Apps > Ring Group

Ring Groups specific default settings.
destination_add_rows numeric 5 TRUE Ring Group “add” rows
default.
destination_edit_rows numeric 1 TRUE Ring Group “edit” rows
default.
8.1.5.18 Security
Security specific default settings.
8.1. Advanced 163

Default Default Default Default Default Setting Description

Setting Setting Setting Setting
Subcate- Name Value Enabled
gory
pass- numeric 15 TRUE Set the required length for the generated passwords.
word_length
pass- boolean TRUE FALSE Set whether to require at least one number in passwords.
word_number
pass- boolean TRUE FALSE Set whether to require at least one uppercase letter in pass-
word_uppercase words.
pass- boolean TRUE FALSE Set whether to require at least one special character in pass-
word_special words.
ses- boolean TRUE TRUE Whether to regenerate the session ID.
sion_rotate
pass- boolean TRUE TRUE Set whether to require at least one lowecase letter in passwords.
word_lowercase
pass- numeric 4 TRUE Set the default strength for generated passwords. Valid Op-
word_strength tions: 1 - Numeric Only, 2 - Include Lower Apha, 3 - Include
Upper Alpha, 4 - Include Special Characters.
8.1.5.19 Server
Server specific default settings.
temp text /tmp TRUE Set the temp directory.
8.1.5.20 Switch
Switch specific default settings. These defaults will change depending if you compiled the SWITCH source or used
the newest default of packages.

de- de- de- de- de-

fault_setting_subcategory
fault_setting_name
fault_setting_value fault_setting_enabled
fault_setting_description
bin dir TRUE Server path for bin.
base dir /usr TRUE Server path for base.
call_center dir /etc/freeswitch/autoload_configs
FALSE Server path for Call Cen-
ter.
conf dir /etc/freeswitch TRUE Server path for Conf files.
db dir /var/lib/freeswitch/db TRUE Server path for sqlite db
files.
dialplan dir /etc/freeswitch/dialplan FALSE Server path for xml di-
alplan
extensions dir /etc/freeswitch/directory FALSE Server path for extension
directory.
grammar dir /usr/share/freeswitch/grammar
TRUE Server path for grammar
xml.
log dir /var/log/freeswitch TRUE Server path for SWITCH
logs.
mod dir /usr/lib/freeswitch/mod TRUE Server path for SWITCH
mod’s.
phrases dir /etc/freeswitch/lang TRUE Server path for SWITCH
xml phrases.
recordings dir /var/lib/freeswitch/recordings
TRUE Server path for SWITCH
recordings.
scripts dir /usr/share/freeswitch/scripts
scripts.
sip_profiles dir /etc/freeswitch/sip_profilesFALSE Server path for SWITCH
xml sip profiles.
sounds dir /usr/share/freeswitch/soundsTRUE Server path for SWITCH
sounds.
storage dir /var/lib/freeswitch/storageTRUE Server path for SWITCH
storage.
voicemail dir /var/lib/freeswitch/storage/voicemail
voicemails.
8.1.5.21 Theme
Theme specific default settings.

background_image array /themes/default/images/backgrounds/blue_blur.jpg
background_color array #6c89b5
background_image_enabled boolean TRUE
logout_icon_visible text FALSE
domain_color text #ffffff
domain_color_hover text #69e5ff
logout_icon_color text #ffffff
logout_icon_color_hover text #69e5ff
menu_main_toggle_color text #ffffff
footer_background_color text rgba(0,0,0,0.1)
footer_color text rgba(255,255,255,0.1)
8.1. Advanced 165

Table 2 – continued from

footer_border_radius text 0 0 4px 4px
message_default_background_color text #fafafa
message_default_color text #666
message_positive_background_color text #ccffcc
message_positive_color text #004200
message_negative_background_color text #ffcdcd
message_negative_color text #670000
message_alert_background_color text #ffe585
message_alert_color text #d66721
message_opacity text 0.9
body_shadow_color text #000000
body_border_radius text 4px
cache boolean FALSE
logo_align text center
menu_main_background_color text #ff0000
menu_main_background_color_hover text #ff0000
menu_main_icons boolean FALSE
menu_main_background_image text /themes/default/images/background_black.png
menu_main_shadow_color text #000000
menu_main_text_color text #ffffff
menu_main_text_color_hover text #69e5ff
menu_main_text_font text Arial
menu_main_text_size text 10.25pt
menu_main_border_size text 1px
menu_main_border_color text #ffffff
menu_position text top
menu_style text fixed
menu_sub_background_color text #000000
menu_sub_icons boolean FALSE
menu_sub_shadow_color text #000000
menu_sub_text_color text #ffffff
menu_sub_text_color_hover text #69e5ff
menu_sub_text_font text Arial
menu_sub_text_size text 10pt
menu_sub_border_radius text 0 0 4px 4px
menu_sub_border_size text 1px
heading_text_font text arial
heading_text_size text 15px
heading_text_color text #952424
body_text_font text arial
body_text_color text #5f5f5f
text_link_color text #004083
text_link_color_hover text #5082ca
table_heading_text_font text arial
table_heading_text_size text 12px
table_heading_background_color text #ffffff
table_heading_border_color text #a4aebf
table_row_text_font text arial
table_row_text_size text 12px


table_row_text_color text #000
table_row_background_color_dark text #e5e9f0
table_row_background_color_medium text #f0f2f6
table_row_border_color text #c5d1e5
dashboard_border_color text #dbe0ea
dashboard_border_color_hover text #cbd3e1
dashboard_border_radius text 5px
dashboard_heading_background_color text #8e96a5
dashboard_heading_background_color_hover text #969dab
dashboard_heading_text_color text #ffffff
dashboard_heading_text_color_hover text #ffffff
dashboard_heading_text_size text 10.5pt
dashboard_heading_text_shadow_color text #000000
dashboard_heading_text_shadow_color_hover text #000000
dashboard_number_background_color text #a4aebf
dashboard_number_background_color_hover text #aeb7c5
dashboard_number_text_color text #ffffff
dashboard_number_text_color_hover text #ffffff
dashboard_number_text_font text Calibri, Candara, Segoe, Segoe UI, Optima, Arial, sans-se
dashboard_number_text_size text 60pt
dashboard_number_text_shadow_color text #737983
dashboard_number_title_text_color text #ffffff
dashboard_number_title_text_font text Calibri, Candara, Segoe, Segoe UI, Optima, Arial, sans-se
dashboard_number_title_text_size text 14px
dashboard_number_title_text_shadow_color text #737983
dashboard_detail_heading_text_size text 11px
dashboard_detail_row_text_size text 11px
dashboard_detail_shadow_color text #737983
dashboard_detail_background_color_center text #f9fbfe
dashboard_footer_background_color_hover text #ebeef3
dashboard_footer_dots_color text #a4aebf
dashboard_footer_dots_color_hover text #a4aebf
form_table_label_padding text 7px 8px
form_table_label_background_color text #e5e9f0
form_table_label_border_color text #ffffff
form_table_label_border_radius text 4px
form_table_label_text_size text 9pt
form_table_label_text_font text Arial
form_table_label_text_color text #000000
form_table_label_required_border_color text #cbcfd5
form_table_label_required_text_color text #000000
form_table_label_required_text_weight text bold
form_table_field_padding text 6px
form_table_field_background_color text #ffffff
form_table_field_border_color text #e5e9f0
form_table_field_border_radius text 0
form_table_field_text_size text 8pt
form_table_field_text_font text Arial
form_table_heading_padding text 8px 8px 4px 8px
8.1. Advanced 167


form_table_row_padding text 3px 0
form_table_row_text_size text 9pt
login_background_color array #6c89b5
login_background_color array #144794
login_background_image_enabled boolean TRUE
login_body_background_color text rgba(255,255,255,0.3)
login_body_shadow_color text rgba(140,140,140,0.3)
login_body_padding text 30px
login_body_width text 100%
login_body_border_size text 1px
login_body_border_color text #ffffff
login_link_text_color text #004083
login_link_text_color_hover text #5082ca
login_link_text_size text 11px
login_link_text_font text Arial
button_background_color_bottom text #000000
button_background_color_hover text #000000
button_background_color_bottom_hover text #000000
button_border_size text 1px
button_border_color text #242424
button_border_color_hover text #000000
button_border_radius text 3px
button_text_size text 11px
button_text_color text #ffffff
button_text_weight text bold
button_padding text 5px 8px
button_height text 28px
input_border_color text #c0c0c0
input_border_color_hover text #c0c0c0
input_border_color_focus text #c0c0c0
input_border_size text 1px
input_border_radius text 3px
input_shadow_inner_color text #cddaf0
input_shadow_outer_color text #ffffff
input_shadow_outer_color_focus text #cddaf0
input_text_size text 12px
input_text_font text Arial
input_text_color text #000000
input_text_placeholder_color text #999999
login_input_background_color text #ffffff
login_input_border_color text #c0c0c0
login_input_border_color_focus text #c0c0c0
login_input_border_size text 1px
login_input_border_radius text 3px
login_input_shadow_inner_color text #cddaf0
login_input_shadow_inner_color_focus text #ffffff
login_input_shadow_outer_color text #ffffff
login_input_shadow_outer_color_focus text #cddaf0
login_input_text_size text 12px


login_input_text_font text Arial
login_input_text_placeholder_color text #999999
font_loader text TRUE
font_loader_version text 1.6.16
font_retrieval text asynchronous
font_source_key text
body_icon_color text rgba(255,255,255,0.25)
body_icon_color_hover text rgba(255,255,255,0.50)
menu_brand_type text image
background_color array #144794
domain_visible text TRUE
menu_main_toggle_color_hover text #69e5ff
message_delay text 1.75
domain_selector_shadow_color text #888888
menu_main_border_radius text 0 0 4px 4px
menu_sub_background_color_hover text
menu_sub_border_color text #ffffff
body_text_size text 12px
table_heading_text_color text #3164ad
table_row_background_color_light text #fff
dashboard_heading_text_font text Calibri, Candara, Segoe, Segoe UI, Optima, Arial, sans-se
dashboard_number_text_shadow_color_hover text #737983
dashboard_detail_background_color_edge text #edf1f7
dashboard_footer_background_color text #e5e9f0
form_table_label_required_background_color text #e5e9f0
form_table_field_text_color text #666666
login_body_border_radius text 4px
button_background_color text #4f4f4f
button_text_font text Candara, Calibri, Segoe, Segoe UI, Optima, Arial, sans-se
button_text_color_hover text #ffffff
input_background_color text #ffffff
input_shadow_inner_color_focus text #ffffff
login_input_border_color_hover text #c0c0c0
login_input_text_color text #000000
body_color text rgba(255,255,255,0.77)
background_image array /themes/default/images/backgrounds/yellowstone_3.jpg
8.1.5.22 Time Conditions
FusionPBX menu Apps > Time Conditions

Time Conditions specific default settings.
8.1. Advanced 169

Default Set- Default Default Setting Value Default Default Setting Descrip-
ting Subcat- Setting Setting tion
egory Name Enabled
region text usa TRUE What region to use by default
when choosing Time Condi-
tions
pre- array {“new_years_day”:{“mday”:”1”,”mon”:”1”}}
TRUE England Holiday
set_england
pre- array {“may_day”:{“mon”:”5”,”mday”:”1- TRUE England Holiday
set_england 7”,”wday”:”2”}}
pre- array {“august_bank_holiday”:{“mon”:”8”,”mday”:”25-
pre- array {“christ- TRUE England Holiday
set_england mas_day”:{“mday”:”25”,”mon”:”12”}}
pre- array {“box- TRUE England Holiday
set_england ing_day”:{“mday”:”26”,”mon”:”12”}}
preset_usa array {“new_years_day”:{“mday”:”1”,”mon”:”1”}}
TRUE USA Holiday
preset_usa array {“presidents_day”:{“wday”:”2”,”mon”:”2”,”mday”:”15-
TRUE USA Holiday
21”}}
preset_usa array {“memorial_day”:{“mday”:”25- TRUE USA Holiday
31”,”wday”:”2”,”mon”:”5”}}
preset_usa array {“indepen- TRUE USA Holiday
dence_day”:{“mday”:”4”,”mon”:”7”}}
preset_usa array {“labor_day”:{“wday”:”2”,”mon”:”9”,”mday”:”1-
TRUE USA Holiday
7”}}
preset_usa array {“columbus_day”:{“wday”:”2”,”mon”:”10”,”mday”:”8-
TRUE USA Holiday
14”}}
preset_usa array {“veter- TRUE USA Holiday
ans_day”:{“mday”:”11”,”mon”:”11”}}
preset_usa array {“black_friday”:{“wday”:”6”,”mon”:”11”,”mday”:”23-
TRUE USA Holiday
29”}}
preset_usa array {“christ- TRUE USA Holiday
mas_day”:{“mday”:”25”,”mon”:”12”}}
preset_canada array {“new_years_day”:{“mday”:”1”,”mon”:”1”}}
TRUE Canada Holiday
preset_canada array {“family_day”:{“wday”:”2”,”mon”:”2”,”mday”:”8-
TRUE Canada Holiday
14”}}
preset_canada array {“victoria_day”:{“wday”:”2”,”mon”:”5”,”mday”:”18-
TRUE Canada Holiday
24”}}
preset_canada array {“canada_day”:{“mday”:”1”,”mon”:”7”}}
TRUE Canada Holiday
preset_canada array {“bc_day”:{“wday”:”2”,”mon”:”8”,”mday”:”1-
TRUE Canada Holiday
7”}}
preset_canada array {“remem- TRUE Canada Holiday
brance_day”:{“mday”:”11”,”mon”:”11”}}
preset_canada array {“christ- TRUE Canada Holiday
mas_day”:{“mday”:”25”,”mon”:”12”}}
preset_canada array {“box- TRUE Canada Holiday
ing_day”:{“mday”:”26”,”mon”:”12”}}
preset_canada array {“labour_day”:{“wday”:”2”,”mon”:”9”,”mday”:”1-
TRUE Canada Holiday
7”}}
pre- array {“spring_bank_holiday”:{“mon”:”5”,”mday”:”25-
preset_usa array {“martin_luther_king_jr_day”:{“wday”:”2”,”mon”:”1”,”mday”:”15-
TRUE USA Holiday
21”}}
preset_usa array {“thanksgiving_day”:{“wday”:”5”,”mon”:”11”,”mday”:”22-
TRUE USA Holiday
170 28”}} Chapter 8. Advanced
preset_canada array {“thanksgiving_day”:{“wday”:”2”,”mon”:”10”,”mday”:”8-
TRUE Canada Holiday
14”}}
8.1.5.23 User
FusionPBX menu Accounts > Users

User specific default settings.
Default Setting Default Set- Default Set- Default Set- Default Setting Description
Subcategory ting Name ting Value ting Enabled
password_special boolean FALSE TRUE Set whether to require at least one special
character in user passwords.
unique text global FALSE Make all user names unique on all do-
mains.
password_length numeric 10 TRUE The default length of characters in a user
password.
pass- boolean TRUE TRUE Set whether to require at least one num-
word_number ber in user passwords.
pass- boolean TRUE TRUE Set whether to require at least one lowe-
word_lowercase case letter in user passwords.
pass- boolean TRUE TRUE Set whether to require at least one upper-
word_uppercase case letter in user passwords.
8.1.5.24 Voicemail
FusionPBX menu Apps > Voicemail

Voicemail specific default settings.
8.1. Advanced 171

Default Setting Default Default Default Default Setting Description

Subcategory Setting Setting Setting
Name Value Enabled
voicemail_file text attach TRUE Define whether to attach voicemail files to email
notifications, or only include a link.
keep_local boolean TRUE TRUE Define whether to keep voicemail files on the
local system after sending attached via email.
storage_type text base64 FALSE Define which storage type (base_64 stores in the
database).
mes- numeric 300 TRUE Maximum length of a voicemail (in seconds).
sage_max_length
pass- numeric 8 TRUE The default length of characters in a voicemail
word_length password.
dis- boolean TRUE FALSE Enable display of @domain_name after voice-
play_domain_name mail_id when rendering emails.
remote_access boolean FALSE TRUE Allow access to the voicemail menu with the
correct voicemail password.
message_order text asc TRUE Set the message order to asc or desc.
pass- boolean TRUE FALSE Enforce voicemail password complexity.
word_complexity
pass- numeric 4 FALSE Minimum voicemail password length.
word_min_length
smtp_from text TRUE SMTP From: specific to Voicemail.
smtp_from_name text TRUE SMTP From: Name specific to Voicemail.
not_found_messageboolean FALSE TRUE Default for not found message.
greet- numeric 90 TRUE Maximum length of a voicemail greeting (in
ing_max_length seconds).
8.1.6 Domains
Welcome to the adding a domain section. Here you will find how to add a domain so that you can reach the specific
tenant from the multi-tenant domain side menu to configure and allow secure administration from the world wide web.
Click here for the youtube video
8.1.6.1 Adding a domain
Control the list of domains to manage.

There are several reasons to create a domain (tenant). One reason would be to organize customers and so customers
have a unique login ie superadmin@domain.tld or superadmin@subdomain.domain.tld as the username.
In this example we will create a domain.
Goto advanced then click Domains.

Then click the
on the right.
This will bring you to enter domain info. (Be sure to create an “A record” from your domain hosting account)
Click save once entry is complete.
8.1. Advanced 173

8.1.6.2 Domain Selection
Changing to a different domain click the stack of three dashes on the top right
A menu will pop open on the right of the screen. Click on the domain that you want to manage. You will always see
the domain you are in by looking at the top right beside the three stacked dashes.
Domains Default Settings
8.1.7 Grammar Editor
Web browser based editor.

8.1.8 Group Manager
Permit access levels to different group of users. The group permissions allow customizing permissions for existing
groups or custom groups.
• superadmin- the global administrator

• admin- the domain administrator
• users- the group for regular users
8.1.8.1 User Manager
Create, edit, remove users.

• Goto Advanced > Group Manager and click USERS at the top right to create, edit or remove a user.
• Click the plus at the right to add a user or pencil to edit an existing user.
8.1. Advanced 175

• Fill in the boxes with pertinent information.

• Group- assign the user to a group. Be wise as to who has access to what.
8.1.9 Menu Manager
Used to customize one or more menus.
8.1.10 Modules
Modules extend the features of the system. Use this page to enable or disable modules.

Modules have several different categories.
8.1.10.1 Applications
8.1.10.2 Auto
8.1.10.3 Dialplan Interfaces
8.1.10.4 Endpoints
8.1.10.5 Event Handlers
8.1.10.6 File Format Interfaces
8.1.10.7 Languages
8.1.10.8 Loggers
8.1.10.9 Say
8.1.10.10 Speech Recognition/Text to Speech
8.1.10.11 Streams/Files
8.1.10.12 XML Interfaces
8.1.11 Number Translations
Use this to translate numbers from the original number to a new number using regular expressions.
8.1. Advanced 177

Activating mod-translate:
• Install the package “freeswitch-mod-translate”. If using Debian Package then use the following command
“apt install freeswitch-mod-translate”
• Configure the module to your likes via the GUI: Advanced -> Number Translations.
• Activate the module in FusionPBX Advanced -> Modules in the Applications section
The documentation for mod-translate can be found under https://freeswitch.org/confluence/display/FREESWITCH/
mod_translate
To use mod-translate to modify inbound calls before they hit the dialplan the following setting for the SIP-profile must be modifi
dialplan “XML” -> dialplan “Translate,XML”
With FreeSwitch 1.8.x it is now possible to specify the translation profile to be used: dialplan “XML” -> dialplan
“Translate:my_profile1,XML”
To activate this setting, the SIP-profile needs to be restarted and the cache flushed.
8.1.12 PHP Editor
An online editor for php specific files for FusionPBX.

8.1.13 Provision Editor
An online editor for phone provisioning templates specific to different vendors for FusionPBX.
8.1.14 Sip Profiles
• Advanced -> SIP Profiles
8.1. Advanced 179

8.1.14.1 Internal
Internal sip profiles (port 5060/5061) require registration or access controls cidr range to allow the IP address in
without SIP authentication. Once the access controls are setup correctly, the carrier will be allowed to send calls to the
internal profile.
8.1.14.2 External
External sip profiles (port 5080-5081) allow anonymous connection to FusionPBX and is optional. External profile is
optional when freewitch has a public ip address. Can be useful when setting behind nat. Being anonymous doesn’t
mean totally open due to the inbound routes call conditions.(call filtering)
8.1.14.3 Internal ipv6
Internal ipv6 sip profiles (port 5060/5061) require registration or access controls cidr range to allow the IP address in
without SIP authentication. Once the access controls are setup correctly, the carrier will be allowed to send calls to the
internal ipv6 profile.
• If you don’t have ipv6 then the ipv6 profiles should be disabled.
• Be sure to stop the profile before disabling it. To disable goto Advanced > SIP Profiles and click the pencil edit
icon to the right of the profile you want to disable. From the dropdown box select enabled to false.
8.1.14.4 External ipv6
External ipv6 sip profiles (port 5080-5081) allow anonymous connection to FusionPBX and is optional.
• If you don’t have ipv6 then the ipv6 profiles should be disabled.
• Be sure to stop the profile before disabling it. To disable goto Advanced > SIP Profiles and click the pencil edit
icon to the right of the profile you want to disable. From the dropdown box select enabled to false.
8.1.15 Scripts Editor
An online editor for script specific files for FusionPBX.

8.1.16 Settings
Switch settings for event socket ip address, event socket port, event socket password, xml rpc http port, xml rpc auth
realm, xml rpc auth user, xml rpc auth password, mod shout decoder, and mod shout volume.
8.1. Advanced 181

8.1.17 Transactions
A list of databse changes (transactions) made by all users while logged into FusionPBX. Changes include
• Domain: The domain the changes occured on.

• User: The user that was logged in at the time the change was made.
• Application: The application that was changed.

• Code: The web server response code.

• IP Address: the ip the user was logged into at the time the change was made.
• Change: The type of change that was made.
• Date: Date the change was made.
Click the edit pencil icon to view more details.
8.1.18 Upgrade
If you are looking to upgrade your current version of FusionPBX to the next release version click here.
The FusionPBX code is constantly evolving.
• Bug fixes being submitted
• Additions to improve security
• Making FusionPBX look nicer
• More flexible
• More scalable
• New features
A complete summary of the changes can be found on the github code page https://github.com/fusionpbx/fusionpbx/
commits/master.
Go to the menu then click on Advanced and then Upgrade. This tool allows you to update the source code, update the
database structure, restore the default menu and permissions. Click here for the Youtube video.
8.1. Advanced 183

Update the source from command line
* cd /var/www/fusionpbx
git pull
chown -R www-data:www-data *
Back to the GUI
*Upgrade Database with advanced -> upgrade schema

*Update permissions
*Update the menu
*Logout and back in
8.1.18.1 How to Upgrade
To upgrade you will need to get the latest source code. Depending on how extreme the changes have been or the
version you currently are on since your last update, you may need to follow version specific upgrade instructions to
bring your install up to date.
Step 1: Update FusionPBX Source
1. GUI -> Advanced -> Upgrade (doesn’t update all files)
Used to update FusionPBX to the latest release.

Upgrade the code via Github/GIT
Login into the web interface with a user account assigned to the superadmin group.
Login to the console with either the ssh, the locally.
Backup It’s a good idea to make a backup. If using sqlite, your backup will easily include the SQL database.
mkdir /etc/fusionpbx
mv /var/www/fusionpbx/resources/config.php /etc/fusionpbx
mv /usr/local/freeswitch/scripts/resources/config.lua /etc/fusionpbx


cd /var/www
cp -R fusionpbx fusionpbx_backup
Change the directory''' to the FusionPBX directory
cd /var/www/fusionpbx
Update the source code (example assumes fusionpbx is in /var/www/fusionpbx)
git pull
Permissions
Reset the permissions on the fusionpbx directory tree. When you do git pull it sets the permissions on any updated
files to match the account that you are running git pull with. If that account is different to the web server account it
will result in some files no longer being accessible and a red bar error at the top of the upgrade screen on the GUI. To
fix this you should reapply the permissions in fusionpbx and recursively in all directories inside it.
The example assumes the web server runs as user ‘www-data’ and fusionpbx is installed to /var/www/fusionpbx.
(chown -Rv Ownername:GroupName /var/www/fusionpbx)
chown -R www-data:www-data *
Step 2: Update Freeswitch Scripts
NOTE: As of FusionPBX 3.8.3 (Stable Branch), the scripts should be automatically updated when updating the
Source Code, using the Advanced > Upgrade page. Any customized scripts, having the same name as the default
scripts, will be overwritten. (An option to disable this default behavior is available using Default Setting: switch >
scripts_update > false) Missing scripts will be restored, and any additional files within the scripts folder will remain
untouched.
FusionPBX is a fast moving project where features are constantly being added and bugs are being fixed on a daily
basis so I would also suggest upgrading the Freeswitch scripts directory as part of any normal upgrade process.
Update Freeswitch
Use github to get the updated files. You have to do this from an empty directory.
cp -R /usr/local/freeswitch/scripts /usr/local/freeswitch/scripts-bak
rm -Rf /usr/local/freeswitch/scripts/
cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx.git
cp -R /var/www/fusionpbx/resources/install/scripts /usr/local/freeswitch
chown -R www-data:www-data /usr/local/freeswitch/scripts
cp -R /usr/local/freeswitch/scripts-bak/resources/config.lua /usr/local/freeswitch/
˓→scripts/resources/config.lua
8.1. Advanced 185

(The last step above is not required if your config.lua file is being stored in a different location, such as the
/etc/fusionpbx folder.)
Clean out this scripts directory

An alternative is to remove the Lua scripts. Only do this if you haven’t customized any LUA scripts
cp -R /usr/local/freeswitch/scripts /usr/local/freeswitch/scripts-bak
rm -rf /usr/local/freeswitch/scripts/*
Pull the most recent scripts down
Here you need to go directly to step 3 and make sure you run upgrade schema from the GUI immediately otherwise
your calls will not complete.
Restore the config.lua file (IMPORTANT!!)
If your config.lua file was located in scripts/resources/, then you’ll need to restore it (from the backup previously
performed) to scripts/resources/config.lua.
Step 3: Upgrade Schema
Many updates have changes to the database and to the Freeswitch scripts. The upgrade_schema script
Upgrade from the GUI
From the GUI, run Advanced -> Upgrade Schema which will add any needed newer tables or columns.
Then run App Defaults. If you removed the scripts on Step 2 then run this twice.

Upgrade from the Command Line

An alternative to running upgrade_schema.php from the GUI is to run the upgrade.php from the command line. It
was designed to make the upgrade easier. If you did not login when updating the FusionPBX source code then you
will need to run the upgrade.php file from the command line. Make sure to use the full path to the PHP file.
As root run the following
/usr/bin/php /var/www/fusionpbx/core/upgrade/upgrade.php
If your screen was nicely formatted with a fusionpbx theme, and suddenly now goes to a black and white screen with
familiar text but no theme, it is because you were using a theme which no longer exists in the latest version of the
code. If this happens to you navigate to:
http://domain_or_ip/mod/users/usersupdate.php
8.1. Advanced 187

Then scroll down to where it says “Template” and select one of the valid templates from the drop down list. Then
press Save. It will be fixed now and you can continue with the remaining steps below.
(Note that any users who have invalid templates selected will also have the same problem you did. You can fix them
from the user manager option in the accounts menu)
Step 4: Apply permissions and Restart Freeswitch
Make sure that the freeswitch directory has the correct permissions
chown -Rv www-data:www-data /usr/local/freeswitch/
Restart Freeswitch
systemctl restart freeswitch
Step 5: Menu
Needed if your menu disappeared.

v1 and v2
Now update the menu to the latest version.
http://domain_or_ip/core/menu/menu_restore_default.php
Press ‘Restore Default’ on the top right.

v3
https://your.ip/core/menu/menu.php
click ‘e’ next to the default menu
click the restore default button.
https://your.ip/logout.php
https://your.ip/login.php
Step 6: Re-generate Settings
Sometimes variable names changes. In rev 1877 v_config_cli.php variable names changed which caused no fax to
email emails or voicemail emails to be sent. Problem was the SMTP details did not exist.
Go to Advanced -> Settings and then click save. This will re-generate v_config_cli.php and any other needs config
files.

8.1.18.2 Move to a different Branch
FusionPBX has a stable and a master(development) branch. You can switch from stable to master but not recomended
to downgrade.
Move to the Stable Branch
mv /var/www/fusionpbx /var/www/fusionpbx-old
cd /var/www && git clone -b 4.4 https://github.com/fusionpbx/fusionpbx.git
chown -R www-data:www-data /var/www/fusionpbx
Make sure config.php exists in /etc/fusionpbx If missing then move it into this directory.
cp /var/www/fusionpbx-master/resources/config.php /etc/fusionpbx
Move to the Master Branch
mv /var/www/fusionpbx /var/www/fusionpbx-old
cd /var/www && git clone https://github.com/fusionpbx/fusionpbx.git
• Complete the normal upgrade process at Advanced -> Upgrade

• If the menu disappears you have to upgrade schema then restore the default menu to get it back.
8.1.19 Variables
Define preprocessor switch variables here.
8.1. Advanced 189

Variables have several different categories.

8.1.19.1 Codecs
8.1.19.2 Defaults
8.1.19.3 IP Address
8.1.19.4 Music on Hold
8.1.19.5 Ringtones
8.1.19.6 Sip
8.1.19.7 Sip Profile:External
8.1.19.8 Sip Profile:Internal
8.1.19.9 Sound
8.1.19.10 Tones
8.1.19.11 Xmpp
8.1.20 XML Editor
An online editor for xml specific files for FusionPBX.
8.1. Advanced 191


CHAPTER 9
Hardware
9.1 Hardware
9.1.1 Auto Provision Phones
Auto provisioning is disabled by default. This is to give a chance to secure provisioning server with HTTP Authen-
tication or CIDR. HTTP Authentication requires the phone to send hash of the combined username and password in
order to get configuration. CIDR is an IP address restriction that can be used to restrict which IP addresses are allowed
to get the device configuration. An example of CIDR is xxx.xxx.xxx.xxx/32 the /32 represents a single IP address.
To set one of these values go to Advanced > Default Settings and find the Provision category from there used the edit
button to set a value. After this is done it is safe to set enabled equal to true.
9.1.1.1 Yealink
To auto provision Yealink

• Login to the phone
• Goto the Security tab at the top right
• On the left vertical menu click Trusted Certificates
• On the dropdown box near the bottom choose Disabled for “Only Accept Trusted Certificates”. If you have a
Certificate that is not self-signed and Approved by Yealink and installed on your FusionPBX server, you can
keep this enabled
• Click Confirm
193
Once you have that done

• Click the Settings tab at the top
• On the left vertical menu click Auto Provision
• Fill in the Server URL field. This will be https://domain.tld/app/provision Replace domain.tld with your actual
domain name
• Click Confirm at the bottom
• Click Auto Provision Now at the bottom
194 Chapter 9. Hardware

9.1.1.2 Polycom
To auto provision Polycom

• Login to the phone. (Default user is admin and default password is 456)
• Go to the top menu an choose Utilities > Phone backup & restore. (Always good to start with factory defaults)
9.1. Hardware 195

• Click The plus to the left of Global Settings then click Restore.

• Login to the phone again.

• Click Settings > Provisioning Server.
9.1. Hardware 197

• Choose the Server Type as http. (If you have ssl certificate that polycom approves then choose https instead.)
• Fill in the Server Address field. This will be domain.tld/app/provision Replace domain.tld with your actual
domain name
• Fill in Server User and Password fields.

• Choose Enable on **Tag SN to UA

• Click Save to Provision the Polycom. You should hear a tone meaning the phone reached out to the server and
provisioned.
9.1.1.3 Cisco SPA
The following information can be used to provisioning Cisco SPA phones.
Basic URL
An example URL for provisioning URL for a Cisco SPA.

http://mydomain.com/app/provision/?mac=\protect\T1\textdollarMA
HTTP Authentication
Phone web interface -> Provision - > Profile Rule
[--uid myUser --pwd myPass]http://mydomain.com/app/provision/?mac=$MA
9.1. Hardware 199

HTTPS
Requires a Cisco Certificate that you will likely need to obtain from a Cisco distributor.
Browser Command
Use your web browser to send the following command to pass the provision the phone now and this will pass URL to
the phone so it has the location neeeded for provisioning the device. In this example 192.168.1.5 is the IP address of
the phone and domain.com needs updated to use the correct tenant domain name.
No HTTP Authentication
http://192.168.1.5/admin/resync?http://domain.com/app/provision/?mac=$MA
With HTTP Authentication
http://192.168.1.4/admin/resync?%5B–uid+admin+–pwd+555%5Dhttp://domain.com/app/provision/?mac=$MA
DHCP Option
Use the DHCP Option 66 to deliver the provisioning URL to the phones without using the web interface.
Additional Information
More information can also be found at https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/

spa100-200/provisioning/guide/SPA100-200_Provisioning.pdf
9.1.1.4 Fanvil
Setting up a Fanvil SIP phone through the phone’s local http management portal.
• Factory reset the phone (physically on the phone) by pressing menu button > Settings > Advanced Settings
(default password is 123) > Reset to Default > Press yes to continue.
• Press Menu > Status to get the phones ip address
• Open a web browser and enter the phones ip address
• Default login name and password is admin
• Top menu click Auto Provision
Common Settings
• Fill out the fields:

– Authentication Name: http user name that was set in FusionPBX default settings
– Authentication Password: http password that was set in FusionPBX default settings
– Save Auto Provision Information: Check the box
– Download CommonConfig enabled: Check the box
– Download DeviceConfig enabled: Check the box

Static Provisioning Server

– Authentication Name: http user name that was set in FusionPBX default settings
– Server Address: https://domain.tld/app/provision
– Protocol Type: HTTPS
– Update Mode: Update after Reboot
• Click Apply
• Left side menu click System

• Top menu click Tools
• Scroll to the bottom and click Reboot
9.1. Hardware 201

Self Signed Certificates
If you are going to use a self signed certificate you will need to adjust additional settings.
• Left side menu click Phone settings
• Top menu click Trusted Certificates
• CA Certificates: Disabled
• Click Apply
9.1.1.5 Grandstream
Auto provisioning with FusionPBX and Grandstream.

• Top Menu > Maintenance > Upgrade and Provisioning.
Fill in the following fields:

• HTTP/HTTPS User Name: Set in Advanced > Default Settings > Provisioning section in your FusionPBX
installation.
• HTTP/HTTPS Password: Set in Advanced > Default Settings > Provisioning section in your FusionPBX
installation.
• Config Server Path: This is typically your FusionPBX installation url/app/provision.
(sub.domain.tld/app/provision)
• Firmware Server Path: Set in Advanced > Default Settings > Provisioning section in your FusionPBX instal-
lation.
• Click Save and Apply at the bottom.
Note: Generally with provisioning, if there is an option like Grandstream has for a box or radio button to choose https
or http then it is not needed to type http:// or https:// in the config url.
• Once you have the proper information filled in, click the Reboot option at the top right.
• Click OK
9.1. Hardware 203

Troubleshooting
• Make sure provisioning is enabled in Advanced > Default Settings

• Check, double check that the correct extension number and password is being used.
• Factory default the phone and try again.
• Reboot the device.
• Check Fail2ban and see if the ip got blocked.
• Make sure you have created an DNS A record for the domain being used and there are no typos
• Nat, firewalls and router settings. Some brands of routers can cause issues. Google the make and model of
router or firewall appliance for common settings or remedies.
• Visit Grandstream Supoprt http://www.grandstream.com/support
9.1.1.6 Htek
Setting up a Htek SIP phone through the phone’s local http management portal.
(default password is admin) > Phone Settings > Factory Reset > Press yes to continue.
• Press Menu > Status > Information to get the phones ip address
• Top menu click Management
• Left menu click Auto Provision
• Fill out the following fields:
– Firmware Server Path:
– Config Server Path:

– HTTP/FTP/HTTPS UserName:
– HTTP/FTP/HTTPS Password:
• Click SaveSet
• Click Autoprovision Now
Self Signed Certificates
Some additional settings need adjusted to provision with a self signed certificate.
• Top menu click Management
• Left menu click Trusted CA
• Choose the following
– Only Accept Trusted Certificates: OFF
– Common Name Validation: OFF
– Trusted Certificates: All Certificates
9.1. Hardware 205

9.1.1.7 Zoiper
QR and app provisioning with Zoiper
This menu add-on will enable the abliity to do QR provisioning from IOS or android Zoiper app. Zoiper has designed
the process in a way that is cross platform. Fusionpbx has the ability to click the extension you want to provision and
a link wil open to either download the app on multiple platforms or if you have the app installed on a mobile device
you can use the QR code scanner to scan a QR image and the mobile is ready to use.
We will walk through the process
Zoiper.com account setup
There are two parts to make this function. http://oem.zoiper.com and Fusionpbx menu add-on.
This all adds a one-click install for both the Desktop and Mobile Zoiper APPs in the User Portal. The page is
accessible by end users.

This can be done with the FREE Zoiper OEM account or can use the paid versions for more customization like
branding.
1. Go to: https://oem.zoiper.com/
2. Sign up for Login
3. Configure your Desktop and Mobile Apps with the information you want.
4. Then click “CONFIGURE” Under Desktop.
5. This will give you a LINK with a PAGE ID:(32 character)
6. https://www.zoiper.com/en/page/MYPAGEID?u=&h=&p=&o=&t=&x=&a=&tr=”
7. Copy the page ID
Zoiper menu add-on for Fusionpbx
On your server
git clone https://github.com/fusionpbx/fusionpbx-apps

cp -R fusionpbx-apps/zoiper /var/www/fusionpbx/app
chown -R www-data:www-data /var/www/fusionpbx/app/zoiper
1. Log into the FusionPBX webpage

2. Advanced -> Upgrade
3. Menu Defaults and Permission Defaults.
4. Log out and back in
Advanced -> Default Settings
Note MYPAGEID and provider_id are two different sets of characters. You can also find these by going into the
oem.zoiper.com login and click “view” on the moblie section.
9.1. Hardware 207

provider_id
provider_id
The Do It Yourself way: make your own page with instructions
If you want to customize this page, you can do so, just make sure to embed this html
˓→code on your website:
<img src="https://oem.zoiper.com/qr.php?provider_id=>>>>>>>
˓→09876543210987654321098765432199<<<<<<<&u=&h=&p=&o=&t=&x=&a=&tr=" alt="QR image" />
MYPAGEID
MYPAGEID
The easy way: send your customers to our landing page
Add a link on your website to this step by step tutorial on our website : (**click
˓→here** to see it in action).
<a href="https://www.zoiper.com/en/page/>>>>>>>>c1234567890123456789012345678901<<<<<<
˓→<?u=&h=&p=&o=&t=&x=&a=&tr=">Configuration instructions for Android and iOS</a>

Goto Advanced -> Default Settings

Add a Default Setting
Category: zoiper
Subcategory: page_id
Type: text
Value: (32 character MYPAGEID)
Enabled: True
Save
Category: zoiper
Subcategory: provider_id
Type: text
Value: (32 character provider_id)
Enabled: True
Save
Goto Apps -> Zoiper

Superadmin will see a list of ALL USER EXTENSIONS
Users will only see the extensions assigned to them.
Click on a link and it will take you to the Zoiper Site. Follow instructions there to download and install.
9.1.2 Manually Provision Phones
How to setup the device using the phone’s web interface.
9.1.2.1 Yealink
Setting up a Yealink SIP phone through the phone’s local http management portal.
• Factory reset the phone by holding down the “OK” button, found in the center of the up/down left/right buttons,
until the phone prompts to Factory reset. Press the softkey for “OK” to continue. The phone will factory reset
and reboot.
• Once the phone powers up, press the OK button once and the phone will display its IP address on the screen.
Navigate to that IP on another device in a web browser, and use admin for the username and admin for the
password.
9.1. Hardware 209

• Enter the following information for each Account that is required:

• Select the “Account” tab at the top menu.
• Set “Line Active” to enabled.
• Set “Label” to the user’s name or the extension number, or both, ie “John-100”.
• Set “Display Name” to the same as above.
• Set “Register Name” to the users FusionPBX extension number, in this case 100.
• Set “User Name” to the users FusionPBX extension number, in this case 100.
• Set “Password” to the users FusionPBX password.
• If you would like to use TCP transport, set “Trasport” to TCP.
• Under SIP Server 1, set the “Server Host” to your FusionPBX domain for that extension.
• Click confirm.

• Click “Advanced” on the left menu.

• Set “Voicemail” to *97
• Click confirm.
9.1.2.2 Polycom
Soundpoint IP 320P
Identification
Display Name: FusionPBX

Address: ExtensionNumber
Authentication User ID: ExtensionNumber
Authentication Password: PASSWORD
Label: ExtensionNumber or Whatever
9.1. Hardware 211


Type: Private
Third Party Name: BLANK
Number Of Line Keys: BLANK
Calls Per Line: BLANK
Server 1
Address: FusionBPX IP ADDRESS or DOMAIN NAME if doing multi-tenant

Port: 5060
Transport: DNSnaptr works or UDP or whatever
Expires: default (30 works ok NATTED)
Register: BLANK
Retry Timeout: BLANK
Retry Maximum Count: BLANK
Line Seize Timeout: BLANK
Server 2
leave alone
Call Diversion
leave alone
Message Center
Subscriber: BLANK
Callback Mode: Contact
Callback Contact: \*98
9.1.2.3 Cisco
To manually provision Cisco

• Login to the phone
• Goto the Ext1 top left tab
• Proxy and Registration section put your servers domain.tld in the proxy field
• Subscriber Information section put the extension number for Display Name and User ID
• Password put the extensions password
• Click Submit All Changes at the bottom

Once you have that done, make sure the p-time is set to 0.020
• Click the advanced option at the top right
• Goto the SIP tab at the top
• Scroll down to the RTP Parameters section and make sure the RTP Packet Size field has 0.020
• Click Submit All Changes at the bottom
HTTP Authentication
Phone web interface -> Provision - > Profile Rule
[–uid myUser –pwd myPass]http://mydomain.com/app/provision/?mac=$MA
9.1.2.4 Fanvil
Setting up a Fanvil SIP phone through the phone’s local http management portal.
(default password is 123) > Reset to Default > Press yes to continue.
9.1. Hardware 213

• Press Menu > Status to get the phones ip address

• Left side menu click Line
– Username:
– Display Name:
– Authentication Name:
– Authentication Password:
– SIP Proxy Server Address:
• Click Apply
• Left side menu click System

• Top menu click Tools
• Scroll to the bottom and click Reboot

9.1.2.5 GrandStream
Registering an Extension using a hardware phone or adapter (ata) using Grandstream.
Granstream is one of the common brand of phone and adapters for voip. From call centers to offices and home offices
Grandstream products can be found. Grandstream has a large selection of hardware from phones, video phones to
analog telephone adapters.
In our example we will register an analog telephone adapter (ata) model HT701.
1. Goto the device ip address. The default password should be admin. Enter admin and click login
9.1. Hardware 215

2. Click on the FXS PORT tab on the top right.
Primary Sip Server: subdomain.domain.com

Failover SIP Server: subdomain1.domain.com (this can be left blank or can use
Primary if only 1 sip server)
SIP User ID: 1000
Authenticated Password: thepassword
Click Update then click Apply at the bottom

3. Click the Status tab on the top left. You should see the Registration as Registered and the User ID 1000
9.1. Hardware 217

• Troubleshooting tips
• Reboot the device.
• Visit Grandstream Supoprt http://www.grandstream.com/support
9.1.2.6 Htek
Setting up a Htek SIP phone through the phone’s local http management portal.
(default password is admin) > Phone Settings > Factory Reset > Press yes to continue.

• Press Menu > Status > Information to get the phones ip address
• Top menu click Account
• Fill out the fields with red* :
– Account Active:
– Primary SIP Server:
– SIP Transport:
– SIP User ID:
– Authenticate ID:
– Authenticate Password:
• Click SaveSet
• Click Restart
9.1. Hardware 219

9.1.2.7 Zoiper
Registering an Extension using the softphone Zoiper.

In the ever changing world of voip businesses are moving away from hardware phones. From call centers to home
offices Zoiper and many other softphones make use of software for communication needs for not only voice but video
and faxing. This example will show how to register an extension using Zoiper for Windows. Note Zoiper can be used
on several operating systems and mobile devices.
1. Download the software. .. Zoiper: http://www.zoiper.com/

2. Install the software.
3. If the software isn’t open click the Zoiper icon to open from the desktop or start menu.
4. Click on Settings
9.1. Hardware 221

5. Click on Preferences

6. Click on Create account
7. Enter the user, password and domain name.
user: 1000
password: thepassword
domain: sub.domain.com
9.1. Hardware 223

8. Click ok. You should have Registered at the top right

• Troubleshooting tips
9.1. Hardware 225

• Visit Zoiper Community Supoprt http://community.zoiper.com/
9.1.2.8 SNOM
From your FusionPBX Install
1. From the Accounts menu, select extensions and select an extension to be provisioned. If no extension exists,
create one.
2. Click the triangle beside Device Provisioning field then enter the mac address of the phone device into MAC
Address field.
3. A barcode scanner can be used or type in the mac address 00041326B92B manually.
4. From the template dropdown list, select the proper make and model of the phone.
5. Click the Save button.
6. The mac address should be a clickable link now. Click that then continue to set up line keys or adjust any other
phone settings.
Note: The provisioning template can be tested by opening up a web browser and entering the provisioning url. The
provisioning url is:
hxxp://voice.example.com/fusionpbx/app/provision/index.php?mac=00041326B92B
Replace the mac address and domain with your own.
From your SNOM phone
Snom like most IP phone has a web admin interface to configure and monitor the phone. Usually, it’s best to be
up-to-date with firmware. If you have issues with provisioning check that the device is up-to-date. Sometimes 1 or 2
versions back are needed also depending on firmware bugs.
1. To find the IP address of the phone, press the menu button on the phone (on the 7XX or 8XX series, or the
settings button on the 3xx series) and press “4” for network then “1” for IP Settings and at the DHCP screen,
press “X” for no. The IP address will appear on the screen.
2. Type the IP address into your web browser and select “Setup > Advanced” from the left menu and select the
“Update” tab from the top.
3. Under “Update Policy”, select “Update Automatically”
4. Under “Setting URL” add in the setting URL as:
hxxp://www.example.com/app/provision/index.php?mac={mac} (Be sure to replace hxxp:// with http://)
The hostname should be replaced with your FusionPBX domain name. Note that we have replaced the domain name
with {mac}. This is a special Snom variable to put the phones Mac address in without having to specify it.

5. Select the “Apply” button and then the “Reboot” button and confirm to reboot the phone.
When the phone reboots, it will be provisioned with your appropriate settings
Using DHCP Option 66 to Deploy the Phone
DHCP is an excellent option for phones deployed in a local office. Your Snom phone can be removed from its box
and simply plugged into the network. All the setting will be retrieved from the server. Be careful to not open up your
FusionPBX to the internet though. Someone who knows your url and a MAC address of a phone can easily retrieve
your phone settings including its password.
Each DHCP Server is different. At Helia we use Cradlepoint MBR 1400 and Cradlepoint MBR 95. Each of these
allow you to setup DHCP option 66. Setting up DHCP directly on the voice server is also an option.
1. On the Cradlepoint MBR 1400 router, select “Network Settings” and ” WiFi / Local Networks”.
2. Select the appropriate “Local IP Networks”, and select the “Edit” button.
3. On the “Local Network Editor” window, select the “DHCP Server” tab
4. Ensure the “DHCP Enable” checkbox is checked and click the “Add” button to add an option.
5. For “option” select “66 Server Name” and for the value, add the provisioning URL:
hxxp://www.example.com/app/provision/index.php?mac={mac} (Be sure to replace hxxp:// with http://)
The hostname should be replaced with your FusionPBX domain name. Note that we have replaced the domain name
with {mac}. This is a special Snom variable to put the phones Mac address in without having to specify it.
With the DHCP information added, the provisioning template will be applied to the phone next time it fetches a new
IP address - usually on its next reboot.
9.1.2.9 Phone Screen Capture
Snom
In order to show the content of the phone display on a computer you need to enter the following URL in a browser:
http://[phoneIP]/screen.bmp
This feature is working on all snom desktop phones. For snom 300 this feature is available in version 8.7.3.7 and later.
Cisco/Linksys SPA 50x and SPA 30x
1. Direct your browser to: http://IP_address_of_phone/admin/screendump.bmp

2. Use the browser to save the file as: anyname.bmp
You now have a 128x64 pixel screen shot in BMP format of you phone's display.
Polycom
Since SIP 3.2.0 you can capture the current screen on a SoundPoint IP, SoundStation IP orVVX phone through the
web interface to the phone.
In order to utilize this facility the Parameter
9.1. Hardware 227

<up up.screenCapture.enabled="1" />

above needs to be added to the Configuration via the Provisioning Server.
Username: Polycom
Password: 456
This does not automatically allow a User to capture the Screen, the functionality needs to be activated by the Phone
User.
Note: You need to re-enable the Screen Capture feature after every phone restart or reboot (repeat below).
Press the Menu Key
Select Settings
Select Basic
Select Preferences
Scroll down and select Screen Capture
Enable or disable the Functionality.
As the browser address, enter http://<phone’s IP address>/captureScreen .
The current screen that is shown on the phone is shown in the browser window. The image can be saved as a file.
Please consult your Admin Guide matching your SIP / UC Software Version.
Yealink
1. Yealink SIP phone with V73 or higher version

2. Login on the WEB interface and fill the Action URI allow IP List (path: Features > Remote Control > Action
URI allow IP List) with any or IP address or your PC, then click Confirm.
1.png
3. In the Brower, fill http://PhoneIP/screencapture in the address bar (Phone IP is the IP address of your phone),
then press Enter key.
4. In the first time, for security consideration, the phone will display a message Allow remote control. Please press
OK. Then repeat step 3.
5. You will get the screen capture of the phone as below:

9.1.2.10 Product Models
SIP-T48G , SIP-T46G , SIP-T42G , SIP-T41P , SIP-T29G , SIP-T28P , SIP-T27P , SIP-T26P , SIP-T23G , SIP-T23P
, SIP-T22P , SIP-T21P E2
9.1.3 Firewall Devices
Firewall device settings that help with SIP connections.
9.1.3.1 ASUS RT-AC66U
This guide was created for the ASUS RT-AC66U router with Firmware Version 3.0.0.4.380_8120. FusionPBX is in
the cloud with a public IP, and the ASUS RT-AC66U router is at the customer’s location with the extensions behind
it. The RT-AC66U is a “prosumer” grade router. It has good performance for the dollar and is a good choice for home
offices.
How to setup QoS
First, enable the QoS feature:

• Log into the router
• Click “Traffic Manager” on the left menu
• Click the “Enable QoS” button to turn it on. Once you do this, the Upload and Download Bandwidth boxes will
appear.
• Enter in your Up/Down speeds. You can use http://beta.speedtest.net/ as noted.
9.1. Hardware 229

• Click the Apply button and wait for the router to reboot.
Next, assign the QoS rules.

• Log back into the router after rebooting, and go to Traffic Manager.
• On the top-right selection box, select “user-defined QoS rules” if it is not already selected
• Your settings may vary based on your environment, but you can use the image below as a good starting point.
• It is important to note that the default rules set “Web Surf” and “HTTPS” to highest priority. We don’t want that
to compete with VOIP traffic, so reduce those to “High.”
• Click Apply.

Note: An important note regarding Priorities

Another important area is the “user-defined priorities” section of Traffic Manager – QoS. As you can see, the default
rules give a very large amount of the bandwidth share to the highest priority. This is very likely excessive for VOIP
traffic. We don’t need much bandwidth, we just need to make sure we get prioritized traffic. You should adjust these
to suit your environment.
9.1. Hardware 231

ASUS RT-AC66U SIP ALG
This guide was created for the ASUS RT-AC66U router with Firmware Version 3.0.0.4.380_8120. FusionPBX is in
the cloud with a public IP, and the ZyXEL USG60 router is at the customer’s location with the extensions behind it.
The RT-AC66U is a “prosumer” grade router. It has good performance for the dollar and is a good choice for home
offices.
How to Disable SIP ALG
• On the left nav menu, click “WAN”
• Click the “NAT Passthrough” tab at the top-right
• Set “SIP Passthrough” to Disable
• Click Apply
• Reboot the router.

This part is a little confusing. It seems that ASUS has either reversed the meaning of SIP Passthrough or changed
how it works over a few firmware releases. At any rate, if you have difficulties with Audio or Registrations, you can
try toggling this setting. With these home-grade routers you should perform a full reboot in order to clear the tables
before testing the phones.
9.1.3.2 Ubiquiti Edgerouter
Ubiquiti Edgerouter Advanced Gigabit Ethernet Router.
9.1. Hardware 233

Port Forwarding
Go to top first menu item Firewall/NAT then second top menu item Port Forwarding.
• Optional: SSH port 22 is optional.

• Required: Sip port range 5060-5090 is recomended.
• Required: HTTPS port 443 is required in order to access your FusionPBX installation and phone provisioning.
• Optional: HTTP port 80 is used by some phone manufactures for provisioning.
• Required: RTP port range 16384-32768.
Note: In order to Port Forward and still have access to the Edgerouter GUI you must change the port number for the
Edgerouter GUI.
Access from another LAN Subnet
If you are behind NAT and are going to use the Edgerouter subnet in addition to an existing subnet (behind another
router) also some setting changes are required. These settings are only recommended in this scenerio.
• Go to First top menu Firewall/NAT tab.

• Go to Second top menu Firewall Policies.

• Edit WAN_LOCAL at the right menu item Actions > Edit RuleSet
• From the Configuration tab, Change the radio button to “Accept” and click “Save Ruleset”.
Warning: Be sure you want to do this and that you are behind either a firewall appliance or another router.
Add Static Route (Double NAT)
This will look different depending on the other router that you might have and what IP range you use.
• A static route is needed on the other router in order for traffic to reach your FusionPBX installation and is only
needed if the Edgerouter is the double NAT.
Scenerio: Router A is the primary router that has a public IP address and a LAN subnet of 10.10.2.1. From this pool of
IP addresses, the Edgerouter gets IP 10.10.2.209. Be sure that router A has DHCP reservation or the ability to make
10.10.2.209 a static IP.
• Router A Router name: This is a label for organizing.
9.1. Hardware 235

• Router A Destination IP address: 192.168.1.38 This is the IP that the Edgerouter gave to your FusionPBX
install.
• Router A Subnet mask: 255.255.255.0 is the subnet mask used in this example.
• Gateway: 10.10.2.209 is the IP Router A gave to the Edgerouter WAN eth0.
• Interface: LAN is a label on Router A to show it’s a local area network address.
Ubiquiti Edgerouter SIP ALG
In some scenerios you may have to turn off SIP ALG.
Check if SIP ALG is running
• Command: lsmod | grep sip
shwim@ubnt:~$ lsmod | grep sip

nf_nat_sip 8853 0
nf_conntrack_sip 21773 1 nf_nat_sip
nf_nat 13284 10 nf_nat_ftp,nf_nat_sip,ipt_MASQUERADE,nf_nat_proto_
˓→gre,nf_nat_h323,nf_nat_ipv4,nf_nat_pptp,nf_nat_tftp,xt_nat,iptable_nat
nf_conntrack 62604 18 nf_nat_ftp,nf_nat_sip,xt_CT,nf_conntrack_proto_gre,

˓→ipt_MASQUERADE,nf_nat,nf_nat_h323,nf_nat_ipv4,nf_nat_pptp,nf_nat_tftp,xt_conntrack,
˓→nf_conntrack_ftp,nf_conntrack_sip,iptable_nat,nf_conntrack_h323,nf_conntrack_ipv4,
˓→nf_conntrack_pptp,nf_conntrack_tftp
shwim@ubnt:~$
This shows that SIP ALG is running in the example above.
Disable SIP ALG
To disable SIP ALG:

• Either click on the CLI button from the Ubiquiti Edgerouter GUI or via you favorite SSH client to the Edgerouter.
• Then type: configure
• Then type: set system conntrack modules sip disable
• Then type: commit
• Then type: save
• Then type: exit

root@ubnt:/home/shwim# configure
[edit]
root@ubnt# set system conntrack modules sip disable
[edit]
root@ubnt# commit
[edit]
root@ubnt# save
Saving configuration to '/config/config.boot'...
Done
[edit]
root@ubnt# exit
Enable SIP ALG
To enable SIP ALG:

• Either click on the CLI button from the Ubiquiti Edgerouter GUI or via you favorite SSH client to the Edgerouter.
• Then type: configure
• Then type: set system conntrack modules sip enable-indirect-media
• Then type: set system conntrack modules sip enable-indirect-signalling
• Then type: commit
• Then type: save
• Then type: exit
root@ubnt:/home/shwim# configure
[edit]
root@ubnt# set system conntrack modules sip enable-indirect-media
[edit]
root@ubnt# set system conntrack modules sip enable-indirect-signalling
[edit]
root@ubnt# commit
[edit]
root@ubnt# save
Saving configuration to '/config/config.boot'...
Done
[edit]
root@ubnt# exit
Note: set system conntrack modules sip port <1-65535> will change the sip port number
9.1.3.3 pfSense
Static Port
Menu -> NAT -> Advanced Outbound NAT (enabled)

set static port to yes
9.1. Hardware 237

Firewall Optimization - Conservative
System -> Advanced -> Firewall NAT -> Firewall Optimization

select Conservative

Create Alias Ports in pfSense
• Configure pfSense to open the necessary ports for FusionPBX and Freeswitch.
• In pfSense navigate to Firewall >> Aliases and click on the Ports TAB.
Name: PBX
Description: FusionPBX
Type: Ports
• Then proceed to add the ports as follows.
Port Description
80 HTTP
443 HTTPS
5060:5061 SIP Internal
5080:5081 SIP External
16384:32768 RTP
• After you are finished Click SAVE.
9.1. Hardware 239

Configure pfSense Port Forwarding
• Click on the ‘+’ to ADD a new Entry.

• Firewall >> NAT >> Port Forward: Add
Interface: WAN
Protocol: TCP/UDP
Destination: <<Select a Public IP from the List>>
Destination Port Range:
from: (Other) PBX
to: (Other) PBX
Redirect target IP: 10.10.0.10

Redirect target port: (Other) PBX
Description: FusionPBX
NAT reflection: Use system default
• Click SAVE when done.

Configure FusionPBX
• In FusionPBX
• System >> Variables > IP Address Section
If you have a static public IP you can replace XX.XX.XX.XX with that IP.
external_rtp_ip XX.XX.XX.XX
external_sip_ip XX.XX.XX.XX
If you have a dynamic IP address you can get a Dynamic DNS from a company such as dyndns.org.
external_rtp_ip myname.dyndns.org
external_sip_ip myname.dyndns.org
Advanced >> SIP Profiles

Edit the Internal Profile and add
Name: aggressive-nat-detection
Value: true
Enabled: True
Status >> SIP Status Stop and Start the internal profile for the changes to take effect.
Note: More information can be found at https://www.netgate.com/docs/pfsense/nat/configuring-nat-for-voip-phones.

html
9.1.3.4 SonicWall TZ-SOHO
This guide was created using 6.5.0.1-14n firmware on a SonicWall TZ-SOHO series UTM router. FusionPBX is in
the cloud with a public IP, and the SonicWall router is at the customer’s location with the extensions behind it.
How to setup Bandwidth Management
First, enable Global Bandwidth Management:

• Log into the SonicWall and go to Security Configuration -> Firewall Settings -> Bandwidth Management
• For Bandwidth Management Type, click the Global radio button.
• This will enable BWM for all traffic.
Enable your required Priority levels. For voice traffic, we’ll enable the “0 Realtime” priority level.
9.1. Hardware 241

The SonicWALL needs to be programmed with your available WAN interface bandwidth. You can go to
beta.speedtest.net or similar to find your speed.
• Log into the SonicWall and go to Network -> Interfaces. Edit your WAN Interface.
• Click the Advanced tab, check both the Egress and Ingress boxes under Bandwidth Management.
• Enter in your speed test values, and click OK

Now create your VOIP services. In this example we’ll use 5060TCP, 5060UDP, and 16384-32768UDP for voice
traffic.
• Go to Policies -> Objects -> Service Objects, and click Add.
• Add objects for your VOIP services. On typical installs this would be 5060TCP/UPD and 16384-32768UDP.
• Click on the Service Groups tab and add all of the services you’ve created to a group.
9.1. Hardware 243

Next, set up an Object for your Cloud PBX:

• Go to Policies -> Objects -> Address Objects, and click Add
• Add your PBX to the WAN Zone assignment with your IP as the Host, or use FQDN if you prefer. If using
multiple servers, add each one and create a group.

Now that we have our Service and Object, we can create a firewall rule and apply prioritization.
• Go to Policies -> Rules -> Access Rules, and click Add.
• Create a rule from the WAN to the LAN, using the VOIP services that you created, and your PBX as the source.
Make sure the Enable SIP Transformation box is unchecked.
• Click the BWM tab and check the Egress and Ingress boxes, with the desired priority level.
9.1. Hardware 245

Save your settings and give it a try!

SonicWall TZ-SOHO SIP ALG
This guide was created for the SonicWall TZ-SOHO router with Firmware Version 6.5.0.1-14n. This has the newer
GUI version and looks quite a bit different than the GUI that had been used in previous years. FusionPBX is in the
cloud with a public IP, and the TZ-SOHO router is at the customer’s location with the extensions behind it.
• Click the MANAGE tab at the top
• On the left menu, go to System Setup-> VOIP
• Check the “Enable consistent NAT” box
• Uncheck the “Enable SIP Transformations” box
• Click ACCEPT
9.1.3.5 ZyXel
This guide was created using V4.2/4.25 firmware on a ZyXEL USG60 series UTM router. FusionPBX is in the cloud
with a public IP, and the ZyXEL USG60 router is at the customer’s location with the extensions behind it.
9.1. Hardware 247

How to setup Bandwidth Management “BWM” aka QoS
There are more than one ways to apply the BWM rules. They can be applied on a service level, or on an object level,
or both. In this example we will provide traffic priority to traffic between the LAN and the cloud PBX.
First, set up an Object for your Cloud PBX.
• Log into the USG and go to Configuration-> Object-> Address/GeoIP
• Click the Add button
Create a name, and enter the static public IP of your FusionPBX. If you have more than one, such as a failover, add
that as well and create a group.
Next, set up a Service Object for the VOIP traffic.

• Go to Configuration-> Object-> Service
• Click the Add button.
Create a name, and set the ports for your traffic. In this example we will add a Service rule for 5060TCP, 5060UDP,
and 16384-32768 UDP.
Note: If you’ve created more than one service object, click the Service Group tab and create a group. Add the service
objects that you’ve created to the group.

Now setup your BWM rules.

• Go to Configuration-> BWM
• Check the Enable BWM box and hit apply.
I’m not sure what affect the “Enable Highest Bandwidth Priority for SIP Traffic” box does, but I leave it unchecked
and it works for me!
• Click the Add button and create a Policy for incoming traffic.
Your settings will vary based on your environment. Priority 1 is the highest priority (what we want) and priority 7 is
the lowest priority.
• Click the Add button and create a Policy for outgoing traffic.
Basically will just switch the Source and Destination.
• Click the Apply button.
9.1. Hardware 249

Zyxel Sip ALG
This guide was created using V4.2/4.25 firmware on a ZyXEL USG60 series UTM router.
Log into the router and navigate to Configuration -> Network -> ALG
Uncheck the following to disable SIP ALG:
• Enable SIP ALG
• Enable SIP Transformations
• Enable Configure SIP Inactivity Timeout
• Restrict Peer to Peer Signaling Connection
• Restrict Peer to Peer Media Connection
Click the Apply button at the bottom of the page. A reboot should not be necessary, but if you’re still experiencing
issues then it is a good idea to try rebooting the router and testing again.

9.1.3.6 Cisco EA6500
This guide was created using a Cisco EA6500 (Linksys AC1750) series router.
Log into the router and navigate to Connectivity -> Administration -> Application Layer Gateway
Uncheck the following to disable SIP ALG:
• Enable SIP ALG
Click the Apply button at the bottom of the page. A reboot should not be necessary, but if you’re still experiencing
issues then it is a good idea to try rebooting the router and testing again.
9.1. Hardware 251


CHAPTER 10
Software
10.1 Software
10.1.1 Software Utilities
There are several software utilities one can use to troubleshoot voip issues and guage quality. Below are a list of some
of the common ones.
10.1.1.1 Packet Capture
tcpdump
Install
apt-get install tcpdump
Command
tcpdump -nq -s 0 -A -vvv -i eth0 port 5060
Tip: you can change the command to suite the proper ethernet device eth0 with what is on your system. Port 5060
can be changed also if you are using a different port.
253
sngrep
Since March 2017 Sngrep is installed on all systems by default. This is a very useful tool to help troubleshoot all types
of sip related issues.
If you installed FusionPBX prior to March 2017 you can still manually install sngrep.
Manual Install
From your FusionPBX install SSH window or console window
cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx-install.sh.git
./sngrep.sh
Command
sngrep
sngrep: https://github.com/irontec/sngrep
10.1.1.2 Call Quality and Monitoring
Call quality can be a nuisance in the voip world. Having a way to track and make reports is a very needed tool.
Homer
Homer is well known to help track and graph quality issues with SIP, like utilizing QoS Reports.
Quote:
HOMER is a robust, carrier-grade, scalable SIP Capture system and VoiP Monitoring Application offer-
ing HEP/EEP, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box,
ready to process & store insane amounts of signaling, logs and statistics with instant search, end-to-end
analysis and drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling
protocol.
To install and configure Homer visit https://github.com/sipcapture/homer
10.1.2 Using SNGREP
10.1.2.1 Main Screen
• Idx: Line number column.

• Method: Type of SIP message column.
• SIP From: SIP message From column.
• SIP To: SIP message To column.
• Msgs: Numerical amount of messages column.
• Source: Source IP and port number column.
254 Chapter 10. Software

• Destination: Destination IP and port number column.

• Call State: Call identifier column.
• ESC Quit: escape and quit sngrep.

• Enter: Show more information about the highlighted line item.
• Space: After pressing the spacebar, the line is selected. With this you can select multiple lines and can be used
with the F2 save option.
• F1 Help: Gives a help menu.
• F2 Save: Option to save the current capture session dialogs to a .pcap or .txt to a specific path and file name.
• F3 Search: Gives the option to search in a more specific and granular way.
• F4 Extended: Gives an extended view.
• F5 Clear: Clear the screen.
• F7 Filter: Like search but with more options to filter the end result.
• F8 Settings: Adjust SNGREP settings interface, capture options, call flow options, and EEP/HEP Homer op-
tions.
• F10: Adjust what columns are displayed on the open sngrep window.
10.1. Software 255

10.1.2.2 SPAM
• User-Agent: Most spam attempts will show an unwanted User-Agent like what is shown in this example.
10.1.2.3 Registration
10.1.2.4 Registration Expanded

10.1.2.5 Call Setup
Invite
200 OK
10.1. Software 257

Call Completed
10.1.2.6 F3 Search
10.1.2.7 F7 Filter

10.1.2.8 F8 Settings
Interface
EEP/HEP HOMER
10.1. Software 259

Call Flow

Capture
10.1. Software 261

10.1.3 TFTP
Several models of phone out there that still only use TFTP for provisioning. Even though they have reached end of life,
some of the popular ones are the Cisco 7960 and 7940. Also would need to add the TFTP port to the server firewall
but this should be allowed only to specific IP addresses as TFTP has no security. Recommend to use TFTP only as a
last resort for phones that don’t support HTTPS.
Install TFTPD
apt-get install tftpd

service xinetd
Change the configuration
edit the /etc/xinetd.d/tftp
Enable TFTP in FusionPBX Gui

Goto Advanced > Default Settings > Provision
Set Enabled to True and define the path to where the TFTP files will be.

Test TFTP
tftp x.x.x.x
get 000000000000.cnf
See the file getting requested for tftp
tail -f /var/log/syslog | grep tftp
10.1. Software 263


CHAPTER 11
Additional Information
11.1 Additional Information
In the Additional Information section you will find topics related to FusionPBX.
11.1.1 Voip Quality
Several factors can attribute to the quality of a Voip call. Most problems with Voip quality can be narrowed down to
packet loss, jitter, wrong configurations, high latency and network attacks.
11.1.1.1 Voip Quality Testing Websites
• https://beta.speedtest.net
• https://www.voipreview.org/speedtest
• https://speedof.me
11.1.1.2 Packet Loss
Packet loss happens when a defined number of packets don’t all reach their destination. Most commonly, this can
happen from faulty network hardware and wiring. Network saturation can be a culprit also on the WAN and LAN of
a network.
0% packet loss is recommended.
11.1.1.3 Jitter
Packets that don’t arrive in the intended order or proper time will result in jitter. This will sound like robotic voice or
missing audio that sounds choppy. Much like a cell phone conversation with poor reception.
265
3 ms in jitter or less is recommended.
11.1.1.4 Latency
Too high of latency will result in conversational timing issues. This sounds like two people talking at the same time.
150 ms or less is recommended.
11.1.1.5 Wrong Configurations
• Quality Of Service QOS when implimented correctly on a network device can help a network provide great Voip
quality.
• ISP provisions your cable modem the wrong speed profile.
11.1.1.6 Network Attacks
We are in the age of the internet wild wild west. Network attacks depending on size can bring a voip call quality
sounding like packet loss, jitter and latency kind of calls.
11.1.2 Freeswitch install
Upgrade Move Source
mv /usr/src/freeswitch freeswitch-version
Git Release
cd /usr/src
git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
cd freeswitch
./bootstrap.sh
or
Git Head
cd /usr/src
git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
cd freeswitch
./bootstrap.sh
266 Chapter 11. Additional Information

or
files.freeswitch.org
cd /usr/src
wget http://files.freeswitch.org/freeswitch-1.4.26.zip
unzip freeswitch-1.4.26.zip
cd freeswitch-1.4.26
1.4.x is considered EOL use the steps below for 1.6.x
cd /usr/src
wget http://files.freeswitch.org/freeswitch-1.6.20.zip
unzip freeswitch-1.6.20.zip
cd freeswitch-1.6.20
Ubuntu Dependencies
apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev

˓→libncurses5-dev libtool make python-dev gawk pkg-config libtiff-dev libperl-dev
˓→libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-
˓→dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev memcached
˓→libmemcached-dev
Debian Dependencies
apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev

˓→libncurses5-dev libtool libtool-bin make python-dev gawk pkg-config libtiff5-dev
˓→libperl-dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-
˓→dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns-dev libpq-dev
˓→memcached libmemcached-dev
CentOS
yum install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-
˓→devel libjpeg-devel openssl-devel e2fsprogs-devel sqlite-devel libcurl-devel pcre-
˓→devel speex-devel ldns-devel libedit-devel libmemcached-devel
Configure services to auto start
chkconfig --add memcached && chkconfig --levels 33 memcached on

chkconfig --add freeswitch && chkconfig --levels 35 freeswitch on
11.1. Additional Information 267

modules.conf
uncomment the FreeSWITCH modules that are needed.
mod_avmd
mod_callcenter
mod_memcache
mod_cidlookup
mod_curl
mod_translate
Used for MP3 support
mod_shout
Postgres driver
./configure --enable-core-pgsql-support
Run Make
make
Remove FreeSWITCH files
This step is only needed for a FreeSWITCH upgrade.

Once it has been confirmed that the compile was successful then remove files from previous version of FreeSWITCH
rm -rf /usr/local/freeswitch/{lib,mod,bin}/*
Install
make install
File Permissions

Set the file permissions instructions may vary based on the OS and install directory.
Debian and Ubuntu
chown -R www-data:www-data /usr/local/freeswitch
CentOS or Other Unix operating systems

(need make sure that the web server has access to IVR recordings, Fax, and Voicemail)
adduser --disabled-password --quiet --system --home /usr/local/freeswitch --gecos

˓→"FreeSWITCH Voice Platform" --ingroup daemon freeswitch
chown -R freeswitch:daemon /usr/local/freeswitch/

chmod -R o-rwx /usr/local/freeswitch/
Install Sound Files
Run this on new installs.
cd /usr/src/freeswitch
make sounds-install moh-install
make hd-sounds-install hd-moh-install
make cd-sounds-install cd-moh-install
Startup Script
Run on new install only. Create the file ‘/etc/init.d/freeswitch’ with the following code:
#!/bin/bash
### BEGIN INIT INFO
# Provides: freeswitch
# Required-Start: $local_fs $remote_fs
# Required-Stop: $local_fs $remote_fs
# Default-Start: 2 3 4 5
# Default-Stop: 0 1 6
# Description: Freeswitch debian init script.
# Author: Matthew Williams
#
### END INIT INFO
# Do NOT "set -e"
# PATH should only include /usr/* if it runs after the mountnfs.sh script
PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/local/bin
DESC="Freeswitch"
NAME=freeswitch


DAEMON=/usr/local/freeswitch/bin/$NAME
DAEMON_ARGS="-nc -nonat -reincarnate"
PIDFILE=/usr/local/freeswitch/run/$NAME.pid
SCRIPTNAME=/etc/init.d/$NAME
FS_USER=www-data #freeswitch
FS_GROUP=www-data #daemon
# Exit if the package is not installed

[ -x "$DAEMON" ] || exit 0
# Read configuration variable file if it is present

[ -r /etc/default/$NAME ] && . /etc/default/$NAME
# Load the VERBOSE setting and other rcS variables

. /lib/init/vars.sh
# Define LSB log_* functions.

# Depend on lsb-base (>= 3.0-6) to ensure that this file is present.
. /lib/lsb/init-functions
#
# Function that sets ulimit values for the daemon
#
do_setlimits() {
ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited
return 0
}
#
# Function that starts the daemon/service
#
do_start()
{
# Set user to run as
if [ $FS_USER ] ; then
DAEMON_ARGS="ècho $DAEMON_ARGS` -u $FS_USER"
fi
# Set group to run as
if [ $FS_GROUP ] ; then
DAEMON_ARGS="ècho $DAEMON_ARGS` -g $FS_GROUP"
fi
# Return
# 0 if daemon has been started
# 1 if daemon was already running
# 2 if daemon could not be started


start-stop-daemon --start --quiet --pidfile $PIDFILE --exec $DAEMON --test > /
˓→dev/null -- \
|| return 1
do_setlimits
start-stop-daemon --start --quiet --pidfile $PIDFILE --exec $DAEMON --
˓→background -- \
$DAEMON_ARGS \
|| return 2
# Add code here, if necessary, that waits for the process to be ready
# to handle requests from services started subsequently which depend
# on this one. As a last resort, sleep for some time.
}
#
# Function that stops the daemon/service
#
do_stop()
{
# Return
# 0 if daemon has been stopped
# 1 if daemon was already stopped
# 2 if daemon could not be stopped
# other if a failure occurred
start-stop-daemon --stop --quiet --retry=TERM/30/KILL/5 --pidfile $PIDFILE --
˓→name $NAME
RETVAL="$?"
[ "$RETVAL" = 2 ] && return 2
# Wait for children to finish too if this is a daemon that forks
# and if the daemon is only ever run from this initscript.
# If the above conditions are not satisfied then add some other code
# that waits for the process to drop all resources that could be
# needed by services started subsequently. A last resort is to
# sleep for some time.
start-stop-daemon --stop --quiet --oknodo --retry=0/30/KILL/5 --exec $DAEMON
[ "$?" = 2 ] && return 2
# Many daemons don't delete their pidfiles when they exit.
rm -f $PIDFILE
return "$RETVAL"
}
#
# Function that sends a SIGHUP to the daemon/service
#
do_reload() {
#
# If the daemon can reload its configuration without
# restarting (for example, when it is sent a SIGHUP),
# then implement that here.
#
start-stop-daemon --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME
return 0
}
case "$1" in
start)
[ "$VERBOSE" != no ] && log_daemon_msg "Starting $DESC" "$NAME"
do_start


case "$?" in
0|1) [ "$VERBOSE" != no ] && log_end_msg 0 ;;
2) [ "$VERBOSE" != no ] && log_end_msg 1 ;;
esac
;;
stop)
[ "$VERBOSE" != no ] && log_daemon_msg "Stopping $DESC" "$NAME"
do_stop
case "$?" in
0|1) [ "$VERBOSE" != no ] && log_end_msg 0 ;;
2) [ "$VERBOSE" != no ] && log_end_msg 1 ;;
esac
;;
status)
status_of_proc -p $PIDFILE $DAEMON $NAME && exit 0 || exit $?
;;
#reload|force-reload)
#
# If do_reload() is not implemented then leave this commented out
# and leave 'force-reload' as an alias for 'restart'.
#
#log_daemon_msg "Reloading $DESC" "$NAME"
#do_reload
#log_end_msg $?
#;;
restart|force-reload)
#
# If the "reload" option is implemented then remove the
# 'force-reload' alias
#
log_daemon_msg "Restarting $DESC" "$NAME"
do_stop
case "$?" in
0|1)
do_start
case "$?" in
0) log_end_msg 0 ;;
1) log_end_msg 1 ;; # Old process is still running
*) log_end_msg 1 ;; # Failed to start
esac
;;
*)
# Failed to stop
log_end_msg 1
;;
esac
;;
*)
#echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2
echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2
exit 3
;;
esac
exit 0
Make the script executable and make it auto start on system boot:

chmod +x /etc/init.d/freeswitch
update-rc.d freeswitch defaults
11.1.2.1 Monit
Used to monitor processes on UNIX systems.

http://mmonit.com/monit/
Install
apt-get install monit
Edit Monit /etc/default/monit and set the “startup” variable to 1 in order to allow monit to start.
Configure
Fail2Ban
cd /etc/monit.d
touch fail2ban
nano fail2ban
Add the following to the file and save it.
check process fail2ban with pidfile /var/run/fail2ban/fail2ban.pid

group services
start program = "/etc/init.d/fail2ban start"
stop program = "/etc/init.d/fail2ban stop"
if 5 restarts within 5 cycles then timeout
SIP
To monitor SIP from local or remote server
cd /etc/monit.d
touch sip
nano sip
check host fusionpbx with address your-ip

if failed port 5060 protocol sip with target monit@monit:5060
then alert

FreeSWITCH
cd /etc/monit/conf.d
or
cd /etc/monit.d
touch freeswitch
nano freeswitch
Add the following

:: #check process freeswitch with pidfile /usr/local/freeswitch/run/freeswitch.pid check process freeswitch with
pidfile /run/freeswitch/freeswitch.pid start program = “/usr/bin/service freeswitch start” stop program =
“/usr/bin/service freeswitch stop”
or
:: #check process freeswitch with pidfile /usr/local/freeswitch/run/freeswitch.pid #start program =
“/usr/local/freeswitch/bin/./freeswitch -nc -u www-data” #stop program = “/usr/local/freeswitch/bin/./freeswitch
-stop” check process freeswitch with pidfile /run/freeswitch/freeswitch.pid start program = “/usr/bin/./freeswitch
-nc -u www-data” stop program = “/usr/bin/./freeswitch -stop”
Additional Options
if 5 restarts within 5 cycles then timeout

if cpu > 60% for 2 cycles then alert
if cpu > 80% for 5 cycles then alert
if totalmem > 2000.0 MB for 5 cycles then restart
if children > 2500 then restart
Monit Daemon Add to the main monit config file.
#monit daemon
set httpd port 2812 and
use address localhost
allow localhost
Monit Commands
monit -h
monit status
11.1.3 SSL/TLS Setup
On a new installation of FusionPBX, TLS for SIP is available to use once you run letsencrypt.sh and make a few
setting changes in FusionPBX.

11.1.3.1 Configure TLS
Configuration for SIP to use TLS can be achieved with the following steps.
• First open an ssh terminal or console window.
• cd /usr/src/fusionpbx-install.sh/debian/resources/
• Execute letsencrypt.sh
• Login to your FusionPBX installation.
• Go to Advanced > Variables.
• Scroll down to SIP Profile: Internal (This can be done on any SIP Profile)
• Set internal_ssl_enable value to true in lowercase.

• Go to Status > SIP Status.
• Click FLUSH CACHE at the top right.
• Click Rescan on the profile.
• You should now see at the right under State (RUNNING)(0)(TLS)

11.1.4 Testimonials
Businesses of all sizes use FusionPBX daily. We love to see folks happy saving money using FusionPBX. Here are
some of the testimonials we received.
I’ve been a longtime VoIP enthusiast for years, since 2005, and I have tried several different hosted/self-hosted PBX
systems. Honestly, FusionPBX wins hands down. What makes it even more amazing is the passion that the FusionPBX
developers and contributors have in their software. I honestly couldn’t be happier with a turn-key PBX system.
-Digital Crisis
Just want to give a thankful shout out to everyone at FusionPBX that have helped in education, contribution and
support. The FusionPBX team have developed a leading product. Its been a joy from day one joining this community
and I look forward to the road ahead.
-Kissvoice
We have been using FusionPBX for many of our clients and just want to express our gratitude to Mark and the team for
not only providing a great product, but being extremely helpful in bringing out new features and helping us maintain
the service. Every new release amazes us with the work and development put into it.
-Kloudphone
SureVoIP
Have been using FusionPBX since 2010. SureVoIP sponsored the first versions of multi-tenant domains and hot
desking. SureVoIP sponsors and contributes fixes and features when possible.
Because of FusionPBX’s highly configurable nature, responsive support team and sane design, SureVoIP have been
able to win many large customers because proprietry systems are so rigid and slow to innovate. We have been proud
to support and deploy FusionPBX for 7 years.
-SureVoIP
Winner of the Best Business ITSP (Medium Enterprise) 2016! http://www.surevoip.co.uk/2016-best-provider
I would like to tell everyone there that I have been trying to get an open source PBX to work for me for over three
months now and since I am not a linux guy, I haven’t been able to get any of them working the way I wanted.
FusionPBX installation script installed ALL required packages and libraries in one go and it was up and running in 10
mins. Once I followed the youtube videos it took me no time to setup and migrate my clients to FusionPBX. One of the
best features I love in FusionPBX is the automatic dialplan expression as I have always struggled with remembering
the expression syntax. The user interface and the way all the features are grouped is awesome.
Again, Thanks for the effort
-BareVOIP Limited

FusionPBX Will Be My Default Choice From Now On.

I just did my first install of FusionPBX a few days ago. All my prior background has been in the Asterisk/FreePBX
community. But my introduction to the freeSWITCH/FusionPBX community has been a very pleasant one. The
change does include a small learning curve, but the advantages far out way the effort required to learn the new platform.
Suddenly a lot of prior headaches and work arounds are gone. The install is painless and simple compared to FreePBX.
And after it is installed it works! This is especially note worthy if you are used to trying to get everything working
with FreePBX on a VPS. This worked perfect on a VPS.
Now for the more practical end of things. I found the features and functionality of FusionPBX to be very comprehen-
sive. The monitoring and control you have over active calls is second to none. And unlike most GUI’s you do not need
to sacrifice functionality for the use of a GUI. With the dial plan manager you can easily add almost any custom dial
plan that you could by editing the xml directly. And as if that is not enough, there is a built in editor for all the xml
and config files.
The support for this project is also noteworthy and it is quite easy to get direct access to the lead developer himself.
Hope that helps. And I highly encourage you to give FusionPBX a try.
Regards,
-ThinkerIV
Amazingly fast to get up and running, but equipped with very powerful functionality as well.
I came from a Trixbox background. I had experienced limitations with the Trixbox solution and was looking for
an alternative when I found FusionPBX. The first thing that amazed me was how simple and speedy it was to get a
working phone system up and running with FusionPBX - far simpler than Trixbox. But then I started to discover how
much advanced functionality was also available, and how extensible the design is - in my opinion it is far easier to
script for FusionPBX than to script for Trixbox if you want to add additional functionality of your own. FusionPBX is
clearly a well thought through design, built on a very solid underlying soft switch (FreeSWITCH) - for me that makes
it the system of choice.
-Stephen
We would love to hear from you! Please reach out to us at http://fusionpbx.com/support.php if you would like to be
featured on this page.
11.1.5 Password Reset
Click here for the new youtube video on password recovery.

The current method to changing the superadmin password is actually to make a new superadmin user name and
password.
Note: In older installations of FusionPBX config.php is located in /var/www/fusionpbx/resources/
1. Move the config.php file temporarily.
cd /etc/fusionpbx
mv config.php config1.php
cat config1.php | grep password

2. Go to the FusionPBX install login page in the web browser. This will put FusionPBX into a recovery mode.
Choose the language for your region and click next.
Note: You will type in your web browser either the ip hxxps://xxx.xxx.xxx.xxx or the domain name
hxxps://sub.domain.tld .
3. In this step, you create what you want for the new superadmin user and password. It has to be a user and
password that does not already exist.

4. Database Host, Database Port, Database name should be pre filled. To provide the Database Username and
Database Password you will have to locate those in the config.php file that we moved eariler. The code block
below shows an easy way to retrieve the database password. Once those are filled in click next.
cd /etc/fusionpbx
cat config1.php | grep password
$db_password = 'databasepasswordfromconfig.php';

5. You should have a new config.php file in the /etc/fusionpbx/ directory. Proceed to login to with the new super-
admin user name and password.
11.1.5.1 Old Password Reset
The steps below are outdated but useful for older installations up to version 4.0. Here are some rough steps to change
the password of the database. The password can only be changed and not recovered.

The database contains a table called v_users which contains the username, password and salt. The password is the
md5 hash of the password and the salt.
11.1.5.2 Password Hash
Use the following commands to generate the password hash. Don’t forget to provide your own salt and password.
echo '<?php $salt = "random-salt-goes-here";$password = "put your password here";

˓→echo md5($salt.$password)."\n"; ?>' > /tmp /test.php
Run the php file from command line.
php /tmp/test.php
11.1.5.3 SQLite
Install sqlite3 which can be be used to modify the database fusionpbx.db. Then open the database with the following:
sqlite3 fusionpbx.db
11.1.5.4 PostgreSQL
Connect to the PostgreSQL database. Once you are running psql you can use:
• \l to list the databases.

• \c to connect to one of them.
• After running the SQL Query then use q to quit.
su postgres
psql
\c fusionpbx
11.1.5.5 Change the Password
The hashed password and the salt can be updated using the command:
update v_users set password = 'replace-with-password-hash-from-php-script', salt =

˓→'replace-with-your-random-salt' where username = 'superadmin';

11.1.6 Feature Codes
Below are the “*” codes used with FusionPBX. You can also create more as needed. If you do be sure to pick ones
that are not currently in use.
11.1.6.1 Basic
Feature Code Name Detail

*1 Call Transfer Transfer a call to another extension
*2 Record Active Call
*4 Attended Call Transfer Attended call transfer to another extension. After extension num-
ber press #
*411 Directory
*DIR to dial by name.
*3472 DISA
*DISA followed by Administrative PIN to receive a dialtone and
call out
Call Privacy Activate call privacy
*67<phone num-
ber>
*69 Call Return Call back the last incoming number
*732 Record
*REC followed by Administrative PIN to record a message
*8[ext] Extension Intercom Page a specific extension.
*870 Redial Redial a number
*9171 Talking Date Current server date
*9170 Talking Time Current server time
*9172 Talking Date & Time Current server data & time
*925 Wakeup Call Schedule a wakeup call
*78 Enable DND Enable Do Not Disturb
*79 Disable DND Disable Do Not Disturb
*9888 FreeSWITCH Confer- Connects to Cluecon Weekly
ence
*0[ext] Speed Dial Speed dial an extension
*21 Follow Me Set the Follow Me number
*72 Enable Call Forward Enables Call Forward
*73 Disable Call Forward Disables Call Forward
*74 Call Forward Toggle Call Forward enable/disable
Administrative PIN (Recordings pin) can be found here Administrative PIN page
11.1.6.2 Call Parking

*5900 Valet Park Attended Transfer (park). The park extension will be played back to you
*5901-5999 Valet Un-Park Retrieve a Valet Parked call

11.1.6.3 Advanced

*8[ext] Extension Intercom Page a specific extension
*33 <ext> Eavesdrop Listen to the call. Press 1 remote, 2 local, 3 full conversation, 0 mute
** <ext> Intercept an extension Intercept a specific extension
11.1.6.4 Voicemail
Feature Name Detail

Code
*97 Voicemail The system detects the extension, and will prompt for your password
*98 Check any Voicemail The system will prompt for both your id (extension number) and pass-
box word
*4000 Check any Voicemail The system will prompt for both your id (extension number) and pass-
box word
*99<exten- Send to Voicemail Send a call directly to voicemail
sion>
11.1.6.5 Miscellaneous

*9192 Info Sends information to the console
*9193 Video Record Record Video
*9194 Video Playback Playback Video
*9195 Delay Echo Audio is played back after a slight delay
*9196 Echo Test Echo Test
*9197 Milliwatt Tone Tone Playback
*9664 Test MoH Test Music on Hold
*You can also add extra feature codes
11.1.7 Features
11.1.7.1 Announcements
Setup a recording for the auto attendant that provides announcement to callers. (See IVR Menus )
11.1.7.2 Authentication
Extendable with plugin support. Web interface authentication by default authenticates against the FusionPBX
Database. LDAP is one and has also been tested with Microsoft Active Directory an OpenLDAP.
11.1.7.3 Call Barge / Eavesdrop / Intercept
Listen into an active call from another extension.

11.1.7.4 Call Block
Block inbound calls by the caller id.
11.1.7.5 Call Broadcast
Create a recording and select one or more groups to have the system call and play the recording.
11.1.7.6 Call Center
Creates a robust call center environment with agent tiers.
11.1.7.7 Call Detail Records
Various reporting capabilities to see who called, when, call length, export to a csv file, and call detail statistics.
11.1.7.8 Call Flows (Day Night Mode)
Typically used with day night mode. To direct calls between two destinations. Can work with BLF on phone to show
which direction call will be directed to.
11.1.7.9 Call Forward
Forward to another extension or to any phone number.
11.1.7.10 Call Monitoring
View which extensions are currently in a call. (see Active Extensions)
11.1.7.11 Call Pickup
For a particular extension or any extension that is currently ringing.
11.1.7.12 Queues
Load calls into queues so they can be answered in the order they came into the queue.
11.1.7.13 Call Recordings
Record all or some calls or parts of the call.
11.1.7.14 Call Routing
Send the call different directions or perform actions based on reading the caller id info or other call information. (see
Dialplan Manager)

11.1.7.15 Call Announced Transfer
Transfer the active call to another internal or external call. Also known as a warm transfer.
11.1.7.16 Call Blind Transfer
Transfer a call like the call was going into a call queue or from an ivr.
11.1.7.17 Call Transfer
Transfer a call.
11.1.7.18 Call Waiting
A beep while on a call and to toggle between two different calls.
11.1.7.19 Caller ID
Support for customization and supporting providers.
11.1.7.20 Conference
Set up voice and video conference calls, is optionally secure with a PIN number, and can transfer current calls to a
conference. Interactive conference control provides ability to see the list of callers in the conference and manage the
volume, see who is talking, kick, mute, unmute, deaf, undeaf, profiles and controls. (See Conference)
11.1.7.21 Conference Center
Unlimited conference rooms with moderator and paticipants, pin numbers, call recording, mute all, caller announce
and more. . .
11.1.7.22 Configuration
While the admin configures the system in the web interface. The data is saved to the database and can optionally be
deliverd to FreeSWITCH via XML files, or on demand from the database.
11.1.7.23 Contacts
Manage your contacts. Import contacts from Outlook CSV files. Export contacts to your cell phone with QR Codes.
It is also possible to add additional features like time cards and invoices that can be related to the contacts.
11.1.7.24 Command
Area to execute commands from the gui. Merged with SQL Query tool with a clip library.

11.1.7.25 Dialplan Manager
The dialplan is used to setup call destinations based on conditions and context. You can use the dialplan to send calls
to gateways, auto attendants, external numbers, to scripts, or any destination.
11.1.7.26 Dial by Name (*411)
Search by first name or last name to find extension numbers on the system.
11.1.7.27 Direct Inward System Access (DISA)
Gives ability to call into the system, put in a pin code, and then call back outbound.
11.1.7.28 Device Provisioning
From Advanced > Default Settings you can enable provisioning for devices. Contacts used as Directory for the phones,
vendor list and functions can be enabled or disabled. Support for memory, expansion (side cars), and programmable
keys. Configure SIP endpoints for Yealink, Polycom, Cisco, Aastra and several other brands.
11.1.7.29 Do Not Disturb (DND)
Direct calls to voicemail by default however there is an option when using do not disturb to send the call to an
alternative destination.
11.1.7.30 Extensions
Create extensions for phones to register to and an option to receive emails on missed calls.
11.1.7.31 Extension Summary
Summary of extension activity per domain such as misssed calls, answered calls, no answer, inbound duration, out-
bound duration, number of outboud calls, number of inbound calls and Average length of Conversation (ALOC). The
summarized information can be downloaded as a CSV file.
11.1.7.32 Editor
File editor for PHP, XML, and Provisioning files.
11.1.7.33 Fax Server
A virtual fax machine that can send and receive faxes with advanced features.
11.1.7.34 Follow Me
Allows calling multiple extensions or external numbers.

11.1.7.35 Gateways
Gateways provide access into other voice networks. These can be voice providers or other systems that require SIP
registration. Check out the Youtube video.
11.1.7.36 Hot Desking
A way to login to another phone device and temporarily or permanently become another extension. This is sometimes
known as ‘hoteling’ and ‘extension mobility’
11.1.7.37 Inbound and Outbound Call Routing
Routes used to receive or send calls in or out of FusionPBX.
11.1.7.38 IVR Menus (Auto Attendant)
Create a structured interactive voice prompt for callers to use. Uses FreeSWITCH IVR and delivered from Database
on Demand. Cached to memcache with IVR Menu Options all editable at once. Also works with Text to Speech.
11.1.7.39 Music on Hold
Allows multiple categories of music on hold that can be set globally or per domain. Can inject additional audio on
intervals such as ‘Your call is very important to us please stand by’.
11.1.7.40 Multi-Tenant
Domain based multi-tenant using subdomains such as red.pbxhosting.tld green.pbxhosting.tld blue.pbxhosting.tld
11.1.7.41 Operator Panel
A virtual panel that agents can drag and drop transfer calls. Adjust call state from available, on break, do not disturb
and logged out.
11.1.7.42 Paging
Page another extension with or without password
11.1.7.43 Parking
Send a call to an unused “park” extension. The caller listens to music on hold until another extension connects to the
call.
11.1.7.44 Phrases
Using xml handler and xml from file system you can string together multiple voice files.

11.1.7.45 Provider Setup
11.1.7.46 Re-branding and Customize
FusionPBX has unprecedented customizability which can be used to meet your needs or the needs of your customers.
Customizable themes, menu, dialplan, and Hundreds of Default Settings to control the theme.
11.1.7.47 Recordings
Create and manage personalized recordings.
11.1.7.48 Ring Groups
Make one extension ring several extensions and an option to receive emails on missed calls.
11.1.7.49 Scalable and Redundant
Can be configured for multi-master database replication, file replication. FusionPBX, Database, and FreeSWITCH
can be distributed across multiple servers for large enterprise scale systems.
11.1.7.50 Time Conditions
A extension that can be timed to route calls based on domain select, global option, move to other domains, and holiday
presets.
11.1.7.51 User and Group Management
Edit, change or add users of all permission levels.
11.1.7.52 Voicemail
Has ability to copy voicemails for other voicemail boxes when receiving a voicemail. Additional features include
voicemail to email and voicemail IVR. Forward add intro, check box for multi-delete.
11.1.7.53 Voicemail to Email
Have voicemails sent to email.
11.1.7.54 Voicemail Transcription
Converts voicemails to text.
11.1.7.55 WebRTC
Make and receive video calls with a web browser.

11.1.7.56 Additional Features
This is not a comprehensive set of features. A complete list would be many times larger. More will be added as time
permits.
11.1.8 Toll Allow
Toll Allow is a variable that can be set per extension. It allows you to limit who can make what type of calls. Note
that although the variable is provided in the extension configuration, the default dialplan DOES NOT make use of it.
Therefore if you want to use it you need to add statements to the dialplan to enable it.
An example for the contents of the toll_allow variable would be:
You can then add information to your dialplan to process this variable. In the example XML below, if the valid allow
value isn’t present then an extension shouldn’t be able to dial out. However extension -> extension should still work.
The following code are example XML for standard outbound routes (Dialplan->OutboundRoutes). Effectively you are
applying an additional condition to EACH outbound route that you want to limit. So in the FusionPBX GUI select an
outbound route and add
condition, type "${toll_allow}", data "local".
Order is important, this should be the FIRST condition of your outbound route.
You’ll need to do that for all of your outbound routes, tag them local, domestic, or interna-
tional depending on what they are. On some installations this example file will be present in
/usr/local/freeswitch/conf/dialplan/default/01_example.com.xml:
11.1.8.1 PERMIT TOLL CALLS
This example assumes all calls are bad (except internal) unless they are flagged as good by the value of the toll_allow
variable.
<include>
<extension name="local.example.com">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{7})$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_
˓→number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}

˓→"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1${default_

˓→areacode}$1"/>
</condition>
</extension>
<extension name="domestic.example.com">
<condition field="${toll_allow}" expression="domestic"/>
<condition field="destination_number" expression="^(\d{11})$">
˓→number}"/>

˓→"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>

</condition>
</extension>


<extension name="international.example.com">
<condition field="${toll_allow}" expression="international"/>
<condition field="destination_number" expression="^(011\d+)$">
˓→number}"/>

˓→"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>

</condition>
</extension>
</include>
11.1.8.2 PREVENT TOLL CALLS
This example takes the opposite approach and is how to PREVENT toll calls. The below example takes the opposite
approach. It assumes that all calls are good unless they are flagged as bad.
Put this in your advanced dialplan. In the toll allow of whatever extension you wanted to restrict put the value ‘local’.
This example restricts from calling 10 or 11 digit numbers.
<extension name="localcalls" >
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="(^\d{10}$|^\d{11}$)">
<action application="hangup"/>
</condition>
</extension>
11.1.9 Network Address Translation
NAT is Network Address Translation. When your FusionPBX and/or FreeSWITCH are inside NAT then then you may
experience one way audio or no audio in either direction the following information can help you get audio working in
both directions.
11.1.9.1 Default config
The external_rtp_ip and external_sip_ip are set to $${local_ip_v4} in Advanced -> Variables by default or Advanced >
Sip Profile settings. The local_ip_v4 variable is auto detected by FreeSWITCH. The variable can be also be overidden
as a preset variable before it is used if you want to control the IP address that it represents.
• This works good when the server has a public IP address.
• It also works well when all phones are inside the same network and nothing needs to traverse the NAT. For
example if you are using a SIP to TDM gateway and all your phones are in the same network.
11.1.9.2 SIP ALG
A SIP Application Layer Gateway (ALG) is a tool designed to help SIP traverse NAT. While the SIP ALG is good in
theory it often causes more problems than it solves. Because of this it’s usually best to disable the SIP ALG on your
firewall. An alternative way to disable it is to move SIP to a non standard port.

11.1.9.3 Static IP
FusionPBX is behind NAT and you have a static public IP address and you have phones on the same network and/or
outside the network.
• Set external_rtp_ip to autonat:xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx can be used also in some instances)
• Set external_sip_ip to autonat:xxx.xxx.xxx.xxx (xxx.xxx.xxx.xxx can be used also in some instances)
• If you don’t register a gateway to the carrier you may need to port forward SIP and RTP.
11.1.9.4 UPnP or PMP
FusionPBX is behind NAT and you don’t have a static ip address. You do have a firewall that is capable of UPnP or
PMP.
• Enable UPnP or PMP in your firewall
• In Debian OS /etc/default/freeswitch remove -nonat
• Make systemd aware of the changes. systemctl daemon-reload
• Set external_rtp_ip to auto-nat
• Set external_sip_ip to auto-nat
• Restart FreeSWITCH. service freeswitch restart
11.1.9.5 Symptoms of misconfigured NAT
• Call drops after 32 seconds.

• One way audio
• No audio
11.1.10 Version Upgrade
Version Upgrade can take several steps to perform. Below will show how to upgrade from specific versions.
11.1.10.1 4.4 to Master (what will become 4.6)
1. Switch branches
mv /var/www/fusionpbx /var/www/fusionpbx-4.4
cd /var/www && git clone https://github.com/fusionpbx/fusionpbx.git
2. Try Advanced -> Upgrade Schema if that fails use the the command line.
php /var/www/fusionpbx/core/upgrade/upgrade.php
3. Refresh the browser if there are issues then logout and then back in.
4. Update the following Dialplans.

If you have made any changes to these make notes on the changes before you delete them. So that the changes could
be added back. For example valet park could have custom music on hold or a custom timeout for the valet park.
user_exists
is_loopback
is_local
user_record
agent_status
group_intercept
extension-to-voicemail
vmain
vmain_user
tone_stream
recordings
valet_park
call_screen
call_forward_not_registered
local_extension
voicemail
• Update these Dialplans by first selecting and deleting their entries from within the Dialplan Manager for all
domains. Then, run Advanced -> Upgrade -> App Defaults to retrieve the new versions of the diaplans.
5. If you have customized any provisioning templates makes sure to copy them from
/var/www/fusionpbx-4.4/resources/templates/provision and copy them into the right vendor directory in
/var/www/fusionpbx/resources/templates/provision. I you haven’t customized the provisioning templates you
can skip this step.
11.1.10.2 Version 4.2 to 4.4
1. Switch branches
mv /var/www/fusionpbx /var/www/fusionpbx-4.2
cd /var/www && git clone -b 4.4 https://github.com/fusionpbx/fusionpbx.git
Note: Depending on when you installed the path /etc/fusionpbx might need created. A good way to tell is once you
move the fusionpbx folder in step one and the FusionPBX is on a page with flags.
**Only** do this step if the folder **doesn't** already exist.
mkdir -p /etc/fusionpbx
mv /var/www/fusionpbx-4.2/resources/config.php /etc/fusionpbx
chown -R www-data:www-data /etc/fusionpbx/
• Then go to Advanced -> Upgrade and update the Source Code, Schema, Menu Defaults and Permission Defaults.
Note: config.lua needs to be read and write by the webserver in order for advanced > default settings to update
config.lua with new path information. Make sure config.lua and config.php are in /etc/fuionpbx/ . Don’t miss this step
chown -R www-data:www-data /etc/fusionpbx/
2. Update the following Dialplans.

user_exists
user_record
call_forward_all
local_extension
• Update these Dialplans by first selecting and deleting their entries from within the Dialplan Manager for all
domains. Then, run Advanced -> Upgrade -> App Defaults to retrieve the new versions of the diaplans.
3. In the menu go to Status then SIP Status and press ‘Flush Cache’.
4. Update old recordings set the record_name and record_path.
cd /usr/src
wget https://raw.githubusercontent.com/fusionpbx/fusionpbx-scripts/master/upgrade/
˓→record_path.php
php record_path.php
5. Resave all Call Center Queues to update each call center queue dialplan. Then restart mod call center or
FreeSWITCH.
6. Advanced > Default Settings
The email section in Advanced > Default settings, changes have been made.
• You will find duplicates with a blank value. The duplicates must be updated with the existing info from the
originals. These duplicates are the new and correct settings. You’ll have to update these blank ones with the
existing values (like smtp server info) to the new default ones. Then delete the original ones.
• Don’t delete the blank entries. The code behind them are for version 4.4+ and the original ones are not.
Note: If you already deleted the blank ones, you’ll have to delete the email section then run Advanced > Upgrade >
App Defaults check box. Then go back to Advanced > Default settings and set the email section back up.
11.1.10.3 Version 4.0 to 4.2
1. Update the source code. From the web interface go to the Menu -> Advanced > Upgrade page. Check the source
box and the press execute. If you see a red bar it indicates there was a git conflict and you will need to update from
console instead. If you don’t see the source box then you will need to update from the console.
git stash
git pull
2. If the page goes blank type in the url http://domain.com/logout.php This should bring you back to the login
screen.
3. Udate the Schema. Advanced -> Upgrade Check the Schema box and then then press execute. https://domain.com/
core/upgrade/index.php
4. Check the box for App Defaults and run execute.
5. Check the box for Menu Defaults and run execute. This will update the menu to the default menu. The menu
should now look like this.

6. Check the box for Permission Defaults and run execute. Permissions are store in a session to get new permissions
logout and back in.
7. Goto Dialplan > Dialplan Manager and delete “local_extension”. Then goto Advanced > Upgrade and only
check box App Defaults and click execute. This will regenerate the new local_extension version.
8. Go to Applications > Conference profiles. Edit each profile and replace $${hold_music} with lo-
cal_stream://default
9. Goto Advanced > Variables hold_music. Make sure it’s value is set as local_stream://default
Check Applications > Music On Hold to see if music is listed properly.

You should see in red default for the category and the kHz sub categories should be
˓→in blue.
If not, do the following
* Edit (Pencil icon on the right) the Category names to reflect default for 8, 16, 32,
˓→ and 48kHz.
* After you click the pencil icon choose at the bottom the domain for the rates and
˓→click save.
* If the category is blank, you may have missed running Advanced > check box app
˓→defaults > execute or you may not have renamed autoload_configs/local_stream.conf.
˓→xml file to local_stream.conf.
* For custom music on hold check the path for the domain name and set select for the
˓→domain name to match the domain used in the path.
10. Remove .xml from the end of the following file names
**Before**
autoload_configs/callcenter.conf.xml
autoload_configs/conference.conf.xml
autoload_configs/local_stream.conf.xml
**After**
autoload_configs/callcenter.conf
autoload_configs/conference.conf
autoload_configs/local_stream.conf
11. Edit autoload_configs/lua.conf.xml adding “languages”. Restart of FreeSWITCH is required.
<param name="xml-handler-bindings" value="configuration,dialplan,directory,languages"/

˓→>
12. Update Time Conditions (Bug Fix)
Goto Advanced > Upgrades page. Check box Update Source, execute.
Goto Advanced > Default settings > Category > delete the category: time condition
˓→presets.
Goto Advanced > Upgrade > check box App Defaults, execute.
Goto Advanced > Default settings. Click "Reload" at the top right. (This will get the
˓→new presets)
Next steps are for existing Time Conditions
Goto Apps > Time Conditions and edit the time conditions remove all holidays and hit
˓→save.
Select the holidays over again.

Note: Many of the provisioning templates were updated. If you use custom provisioning templates you should
consider updating them with the new versions.
11.1.10.4 Version 3.8 to 4.0
Remove the comments from the script-directory in /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

If using the FreesWITCH package then remove $${base_dir} and set the full path to the scripts directory.
before: 
after: <param name="script-directory" value="/usr/local/freeswitch/scripts/?.lua"/>
Rebooting FreeSWITCH is required for this to take effect.
11.1.10.5 Version 3.6 to 3.8
Note: Upgrading can get very complex. If the production system is critical or you are intimidated from these
upgrade instructions you may want FusionPBX paid support at http://www.fusionpbx.com/support.php
A standard ‘upgrade’ procedure should always be followed:

(1. Make a Backup!, 2. Advanced > Upgrade steps, 3. Update switch scripts, 4. Restart FreeSWITCH).
Beyond the standard upgrade procedure just described, the following will also need to be performed:
uncomment: <param name="script-directory" value="$${base_dir}/scripts/?.lua"/>

in: /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml
* Rebuild all time conditions.

* After you edit a particular time condition, click the Dialplan button on the top right to see what was there originally.
* Delete the following dialplans from each domain then run Advanced -> Upgrade -> App Defaults. If using XML
handler for the dialplan flush memcache. If using dialplans XML on the file system resave one of the dialplans to
have FusionPBX rewrite the XML files.
* user_exists - call_timeout variable was added
* extension-intercom - It has been renamed to ‘page-extension’
* eavesdrop - Change ‘*‘88[ext] to ‘*‘33[ext] so that it doesn’t conflict with page-extension at ‘*‘8[ext]
* user_status - Has been renamed to ‘agent_status’
* page - Dialplan has been simplified.
* valet_park_out - Changed regex variable from $1 to $2
* local_extension - failure handler was added to support call forward on busy and no answer
* If using call center feature code ‘*‘22 edit each agent and add an agent id and password (pin number)
* Delete any dialplan with the ‘features’ context. These have been moved into the dialplan domain contexts.
* If using App -> XMPP, Content Manager, or Schema they have been moved dev -> branches -> apps directory need
to pull files from there if you want to use any of them.
* For single tenant systems ‘default’ context is no longer used by default.
* Easiest way to update your system is go to Advanced -> Domains and edit your domain.

* Copy your current domain name then change the name to default then save the change.
* Now edit the domain name again and paste your original domain name or IP address whatever the domain
originally was and save the changes
* Go to accounts extensions and save one extension. (not needed if using the XML handler)
* Go to Dialplan Manager and save one of the dialplans. (not needed if using the XML handler)
* FAX ( may require adjusting the paths and web server user account to match your server ‘www-data’ is used in this
example)
* Delete all previous FAX dialplans
* Resave each fax server in the GUI.
* cd /var/www/fusionpbx/app/fax
* wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/fax_import.php
* chown -R www-data:www-data fax_import.php
* Login into the GUI and use this path in your browser http://<domain-or-ip>/app/fax/fax_import.php
* rm /var/www/fusionpbx/app/fax/fax_import.php
* Groups and Permissions
If you go to Advanced Group Manager -> And you see what looks like duplicates of user, admin and superadmin
groups then you need do the following instructions.
Remove permissions associated with all domain groups with names that match default global groups. . .
Use the Advanced -> SQL Query tool to do the following.
delete from v_group_permissions where domain_uuid is not null

and (
group_name = 'user'
or group_name = 'admin'
or group_name = 'superadmin'
or group_name = 'agent'
or group_name = 'public'
)
Remove all domain groups having the same names as the default global groups
(retains any custom domain groups)...
delete from v_groups where

domain_uuid is not null
and (
group_name = 'user'
)
Empty the group_uuid field for any group user with a group_name value having
the same name as the default global groups (retains user assignments to custom domain
˓→groups)...

update v_group_users set group_uuid = null where

group_name = 'user'
For group users with a null group_uuid, insert the group_uuid of the global group that matches the group_name
value. . .
Run this code from Advanced -> Command -> PHP Command.
$sql = "select group_user_uuid, group_name ";

$sql .= "from v_group_users where group_uuid is null";
$prep_statement = $db->prepare(check_sql($sql));
$prep_statement->execute();
$result = $prep_statement->fetchAll(PDO::FETCH_NAMED);
$result_count = count($result);
unset($prep_statement);
if ($result_count > 0) {
foreach($result as $field) {
//note group user uuid
$group_user_uuid = $field['group_user_uuid'];
$group_name = $field['group_name'];
//get global group uuid
$sql = "select group_uuid from v_groups ";
$sql .= "where domain_uuid is null ";
$sql .= "and group_name = '".$group_name."' ";
$prep_statement = $db->prepare($sql);
$prep_statement->execute();
$sub_result = $prep_statement->fetch(PDO::FETCH_ASSOC);
$sub_result_count = count($sub_result);
unset ($prep_statement);
//set group uuid
if ($sub_result_count > 0) {
$sql = "update v_group_users ";
$sql .= "set group_uuid = '".$sub_result['group_uuid']."' ";
$sql .= "where group_user_uuid = '".$group_user_uuid."' ";
$count = $db->exec(check_sql($sql));
unset($sql);
}
}
}
Apps menu disappeared
If your apps menu disappeared check that it wasn’t set to protected in the menu manager.
(advanced -> menu manager). If protected is true, it won’t show up.

11.1.10.6 Version 3.5 to 3.6
When running Upgrade -> Schema

If you see ALTER TABLE v_xml_cdr ADD json json; every time you run the upgrade schema then you likely have
an old version of Postgres. To fix this either upgrade to the latest Postgres server or run the following SQL statement
from advanced -> sql query.
ALTER TABLE v_xml_cdr ADD json text;
See https://github.com/fusionpbx/fusionpbx/issues/655 for more details.
Potential issue with call recording after upgrading/switch to latest 3.6 stable.
After upgrading to 3.6 stable from 3.5 dev I noticed that calls were no longer being recorded. This was due to the file
extension being missing from the recording path. If this is happening to you it is an easy fix.
Go to Advanced -> variables -> category default and add the variable record_ext and set it to either wav or mp3.
Choosing mp3 depends upon whether or not you have mod_shout installed and enabled.
11.1.10.7 Version 3.4 to 3.5
Gateways now use the gateway_uuid as the name that is used when interacting with FreeSWITCH. This script is
needed to help change the gateway names used in the outbound routes. You may need to remove the old gateway file
names from the conf/sip_profiles/external directory.
wget http://fusionpbx.googlecode.com/svn/branches/dev/scripts/upgrade/gateway_uuid.php
http://x.x.x.x/gateway_uuid.php
rm gateway_uuid.php
* Go To Advanced -> Default Settings -> Switch Category -> Sub category gateways change to sip_profiles
Permissions Issues - (access denied errors)

Due to changes which improve consistency throughout the product, some Users have had problem with superadmin
receiving “access denied” errors after the upgrade.

* Go To Advanced -> Group Manager

* On superadmin click Permissions and then Restore Default
You may need to execute this operation for each group.
Default Settings’
In the switch category change gateways to sip_profiles
11.1.10.8 Version 3.3 to 3.4
Update the source as described on this page, menu manager restore default, group manager edit a group restore
default, advanced -> upgrade schema.
FusionPBX 3.4 hunt groups have been deprecated. Use the following script run it only one time to move existing
hunt groups to ring groups.
wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/hunt_group_
˓→export.php
http://x.x.x.x/hunt_group_export.php
rm -r hunt_group_export.php
Ring groups were expanded to add ability to call external numbers and match other missing hunt group features. A
new table was created to accomodate this.
wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/ring_group_
˓→extensions.php
http://x.x.x.x/ring_group_extensions.php
rm ring_group_extensions.php

11.1.10.9 Version 3.2 to 3.3
FreeSWITCH changed the syntax to connect to the database so numerous LUA scripts had to be updated. If you
customized any of the lua scripts make a backup of the FreeSWITCH scripts directory. Then remove the contents of
the freeswitch/scripts directory and then run advanced -> upgrade schema (which will detect the missing scripts
and replace them).
11.1.10.10 Version 3.1.4 to 3.2
Ubuntu/Debian
git pull
Advanced -> Upgrade Schema
Menu
If you cant see the menu after upgrading try the following in your browser replace x.x.x.x with your ip or domain
name.
x.x.x.x/core/menu/menu.php
Edit the menu make sure the language is set to en-us.
Press **Restore Default**
Default settings
x.x.x.x/core/default_settings/default_settings.php
category: language
type: code
value: en-us
Email
Migrating email to the new FusionPBX native voicemail.
wget https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade/voicemail_
˓→export.php

Run from the browser it will take the voicemail data from the FreeSWITCH database and copy the information into
the FusionPBX database.
http://x.x.x.x/voicemail_export.php
Remove the export file
rm voicemail_export.php
Call Forward / Follow Me
No longer using hunt groups. So the backend has changed so keep in mind that you need to reset call forward and
follow me settings. They are still listed in app -> hunt groups. After updating the info in call forward, follow me
you should delete the hunt group.
11.1.10.11 Version 2 to 3.0
LESS than or EQUAL to revision 1877, use the migration tool.

https://github.com/fusionpbx/fusionpbx-scripts/tree/master/upgrade
If greater than revision 1877, use latest.
| When upgrading from previous versions, you may encounter the following issues:
Changes to your dial plan or extensions don’t take effect

* Go to the Advanced -> Default Settings page
* Remove “/default” from the end of your dialplan and extensions directories
Missing menus
* Go to hxxps://yourdomain.com/core/menu/menu.php
* Click the edit (e) button beside default
* Click the Restore Default button
* Check that all the entries in the list are accessible by the appropriate groups
Emails not being sent for voicemail or fax

* Double check the SMTP settings on the System -> Settings page
* Save it, even if you haven’t changed anything

Release Revisions
• r0001 is 1.0 release - 6 Nov 2009
• r2523 is 3.0 release - 3 May 2012
• r2585 is 3.0.4 release - 24 May 2012
• r2757 is 3.1 release - 18 Aug 2012
• r2777 is 3.1.1 release - 26 Aug 2012
• r2827 is 3.1.2 release - 12 Sep 2012
• r2897 is 3.1.3 release - 26 Sep 2012
• r2907 is 3.1.4 release - 27 Sep 2012
• r3694 is 3.2 release - 19 Jan 2013
• r3978 is 3.3 release - 1 May 2013
• r4605 is 3.4 release - 28 Sep 2013
• r6747 is 3.6.1 release - 22 Aug 2014
• r8481 is 3.8.3 release - 11 May 2014
• r793d386 is 4.0 release - Aug 2015
• r4fdb6e9 is 4.1 release - Dec 2015
• rxxxxxxx is 4.2 release - xxx 2016
11.1.10.12 SQLite
SQLite is the FreeSWITCH default. Databases are located in the freeswitch/db directory.
11.1.10.13 ODBC
http://wiki.freeswitch.org/wiki/ODBC
11.1.10.14 Postgres
Postgres native support will be in FreeSWITCH 1.2.4 but has been available in the Main GIT branch.
11.1.10.15 Dependencies
libpq and the associated dev packages are required
11.1.10.16 Configure
To enable PostgresSQL as a native client in FreeSWITCH you must enable it during the build when running configure.
** ./configure –enable-core-pgsql-support **

11.1.10.17 switch.conf.xml
Under the Settings area insert the following line

<param name=”core-db-dsn” value=”pgsql;hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch
password=” options=’-c client_min_messages=NOTICE’ application_name=’freeswitch’” />
11.1.11 Releases
GIT Version Date

4.4.1 28 May 2018
4.4.0 5 April 2018
4.2.2 30 January 2017
4.2.1 14 December 2016
4.2.0 11 September 2016
4.0.0 16 August 2015
SVN Version Date

r6404 3.6 Aug 17, 2014
r6846 3.6.2 Sept 11, 2014
r7280 3.6.3 Nov 24, 2014
r8427 3.8 May 5, 2015
11.1.12 Regular Expressions
• ^ Start of the string

• $ End of the string
• ? optional example 1? makes the 1 optional
• \d{10} 10 digits
• ( and ) gets matching digits inside brackets sets a $1 and second set of brackets creates $2
• ^\+?1?(\d{10})$ 10 to 11 digits and e164 format sets $1 to 10 digits
• [0-9] Any number between 0 to 9
• [2-9] Any number between 2 to 9
• | The pipe works like an OR. Example ^101$|^102$ matches 101 or 102
• ^9(\d{10})$ This strips off the 9 and the $1 value is the remaining 10 digits
Dialplan Expression
• Two digits: ^(\d{2})$
• Three digits: ^(\d{3})$
• Four digits: ^(\d{4})$
• FIve digits: ^(\d{5})$
• Six digits: ^(\d{6})$
• Seven digits(Local Calling): ^(\d{7})$

• Eight digits: ^(\d{8})$

• Nine digits: ^(\d{9})$
• Ten digits(Long Distance): ^(\d{10})$
• Eleven digits(Long Distance with a 1): ^\+?(\d{11})$
• North America: ^\+?1?(\d{10})$
• North America International: ^(011\d{9,17})$
• Caribbean: ^(?:+1|1)((?:684|264|268|242|246|441|284|345|767|809|829|849|473|876|664|670|787|939|869|758|784|721|868|649)\d
• Europe International: ^(00\d{9,17})$
• International: ^(\d{12,20})$
• 311 Information: ^(311)$
• 711 TTY: ^(711)$
• 911 Emergency: ^(911)$
• Toll Free: ^1?(8(00|55|66|77|88)[2-9]\d{6})$
• INUM: ^0118835100\d{8}$
• Dial 9 then Two digits: ^9(\d{2})$
• Dial 9 then Three digits: ^9(\d{3})$
• Dial 9 then Four digits: ^9(\d{4})$
• Dial 9 then Five digits: ^9(\d{5})$
• Dial 9 then Six digits: ^9(\d{6})$
• Dial 9 then Seven digits: ^9(\d{7})$
• Dial 9 then Eight digits: ^9(\d{8})$
• Dial 9 then Nine digits: ^9(\d{9})$
• Dial 9 then Ten digits: ^9(\d{10})$
• Dial 9 then Eleven digits: ^9(\d{11})$
• Dial 9 then International: ^9(\d{12,20})$
Links
• https://regex101.com/
• https://regex101.com/r/QmOZiH/3/
• https://regexr.com/
• https://extendsclass.com/regex-tester.html
11.1.13 PostgreSQL
PostgreSQL is a enterprise grade open source database. http://www.postgresql.org/

11.1.13.1 Backup
The following assumes the database username is fusionpbx and the database to backup is fusionpbx. Make sure you
have the database password ready.
su postgres
pg_dump -U fusionpbx fusionpbx -b -v -f /tmp/fusionpbx.sql
11.1.13.2 Restore
Assuming username fusionpbx and database fusionpbx

psql -U fusionpbx -d fusionpbx -f fusionpbx.sql
11.1.13.3 Console
su postgres
psql
list the databases

\l
connect to the database
\connect fusionpbx
or
\c fusionpbx
list tables
\d
drop the database
DROP DATABASE fusionpbx;
create the database
CREATE DATABASE fusionpbx;
11.1.13.4 Links
http://www.mkyong.com/database/backup-restore-database-in-postgresql-pg_dumppg_restore/
http://www.postgresql.org/docs/9.1/static/backup.html
11.1.14 Shared Line Appearance
Shared Line Apprearance(SLA) also known in older phone systems as a “Key System”.

11.1.14.1 FusionPBX Settings
FusionPBX Menu Advanced > SIP Profiles > edit a profile to enable SLA.
• name: multiple-registrations
• value: contact
• enabled: true
11.1.14.2 Yealink SLA
FusionPBX Menu Accounts > Device > Edit a specific device.
• Line > Shared Line

• Options: 0-Disabled (default), 1-Broadsoft SCA, 2-BLA
11.1.14.3 Polycom SLA
FusionPBX Menu Accounts > Device > Edit a specific device.
• Line > Shared Line

• Options: shared, private
11.1.15 CDR Archive Server
• Note: This feature is on version 4.5+ and requires PostgreSQL ver 9.5+
Fusionpbx has the ability to access CDR records on a seperate archive database. This is helpful for longterm CDR
storage while keeping your active database small. When the feature is enabled you will see an “ARCHIVE” button in
CDR page that accesses records on your archive database.
The first step is to install an archive database. This can be done by standing up another fusionpbx server or by setting
up a postgres server. If postgres is installed by itself you will need to manage the indexes, tables names and column
names manually on the archive server. They need to match the values on the live database.
Once you get your archive database setup and can access both databases in both directions (live <-> archive), you will
need a mechanism to move the CDR Records from the live database to the archive database. In this example I have a

complete fusionpbx install on the archive server. That way I can use the fusionpbx web gui to explore the records and
use the Upgrade feature to keep the table & column names in sync.
11.1.15.1 Move the Records
Create a shell script to copy the records.
touch /etc/cron.daily/db_copy.sh
chmod +x /etc/cron.daily/db_copy.sh
nano /etc/cron.daily/db_copy.sh
#!/bin/sh
#copy the data from the fusion db to a local csv file

psql --host=x.x.x --username=fusionpbx -c "\copy (SELECT * FROM v_domains) TO '/tmp/
˓→domains.csv' WITH CSV"
psql --host=x.x.x --username=fusionpbx -c "\copy (SELECT * FROM v_fax_logs) TO '/tmp/

˓→fax_logs.csv' WITH CSV"
psql --host=x.x.x --username=fusionpbx -c "\copy (SELECT * FROM v_xml_cdr) TO '/tmp/

˓→xml_cdr.csv' WITH CSV"
psql --host=x.x.x --username=fusionpbx -c "\copy (SELECT * FROM v_conference_

˓→sessions) TO '/tmp/conference_sessions.csv' WITH CSV"
psql --host=x.x.x --username=fusionpbx -c "\copy (SELECT * FROM v_conference_session_

˓→details) TO '/tmp/conference_session_details.csv' WITH CSV"
#Insert the data into the cdr server

# - create a temp tables
# - copy the csv data to the temp tables
# - insert data from the temp table to the real tables
# - delete the temp tables
# - remove the json data from the cdrs. too much space
psql --host=x.x.x.x --username=fusionpbx << EOF
CREATE TEMP TABLE tmp_domains AS SELECT * FROM v_domains WITH NO DATA;

CREATE TEMP TABLE tmp_fax_logs AS SELECT * FROM v_fax_logs WITH NO DATA;
CREATE TEMP TABLE tmp_xml_cdr AS SELECT * FROM v_xml_cdr WITH NO DATA;
CREATE TEMP TABLE tmp_conference_sessions AS SELECT * FROM v_conference_sessions WITH
˓→NO DATA;
CREATE TEMP TABLE tmp_conference_session_details AS SELECT * FROM v_conference_

˓→session_details WITH NO DATA;
COPY tmp_domains FROM '/tmp/domains.csv' DELIMITER ',' CSV HEADER;

COPY tmp_fax_logs FROM '/tmp/fax_logs.csv' DELIMITER ',' CSV HEADER;
COPY tmp_xml_cdr FROM '/tmp/xml_cdr.csv' DELIMITER ',' CSV HEADER;
COPY tmp_conference_sessions FROM '/tmp/conference_sessions.csv' DELIMITER ',' CSV
˓→HEADER;
COPY tmp_conference_session_details FROM '/tmp/conference_session_details.csv'

˓→DELIMITER ',' CSV HEADER;
INSERT INTO v_domains SELECT DISTINCT ON (domain_uuid) * FROM tmp_domains ON CONFLICT

˓→DO NOTHING; (continues on next page)


INSERT INTO v_fax_logs SELECT DISTINCT ON (fax_log_uuid) * FROM tmp_fax_logs ON
˓→CONFLICT DO NOTHING;
INSERT INTO v_xml_cdr SELECT DISTINCT ON (xml_cdr_uuid) * FROM tmp_xml_cdr ON

˓→CONFLICT DO NOTHING;
INSERT INTO v_conference_sessions SELECT DISTINCT ON (conference_session_uuid) * FROM

˓→tmp_conference_sessions ON CONFLICT DO NOTHING;
INSERT INTO v_conference_session_details SELECT DISTINCT ON (conference_session_

˓→detail_uuid) * FROM tmp_conference_session_details ON CONFLICT DO NOTHING;
DROP TABLE tmp_domains;

DROP TABLE tmp_fax_logs;
DROP TABLE tmp_xml_cdr;
DROP TABLE tmp_conference_sessions;
DROP TABLE tmp_conference_session_details;
UPDATE v_xml_cdr SET json = NULL;
EOF
#remove the csv files

rm /tmp/domains.csv
rm /tmp/fax_logs.csv
rm /tmp/xml_cdr.csv
rm /tmp/conference_sessions.csv
rm /tmp/conference_session_details.csv
Add to cron
crontab -e
15 0 * * * bash /etc/cron.daily/db_copy.sh
• Note: In this example I remove the json data from the records. You will need to comment out the “SET json =
NULL” line if you want to keep the call variables.
CDR
FusionPBX menu Apps > CDR

Setup your live server to connect to the archive database.

Default Setting Sub- Default Setting Default Setting Setting En- Default Setting Description
category Name Value abled
archive_database_driver text pgsql TRUE Archive Database Driver
archive_database_host text x.x.x.x TRUE IP/Hostname of Archive
Database
archive_database_password
text somethingSecret TRUE Archive Database Password
archive_database_port text 5432 TRUE Archive Database Port
archive_database_username
text fusionpbx TRUE Archive Database Username
archive_database boolean TRUE FALSE Enable Dedicated CDR
Database Access
archive_database_name text fusionpbx FALSE Archive Database Name


CHAPTER 12
Contributing
12.1 Contributing
There are many ways to help the FusionPBX project.

What We Need:
1. Developers
2. Technical Writers
3. Translators
4. Quality Assurance Testers
5. Documentors
Note: If you are planning to contribute to any of our github repos we require that you sign the FusionPBX Contributor
License Agreement. This mainly protects FusionPBX and our users read: you from code that could be inserted that
might pose a legal problem. It does this by verifying that the code you are contributing is yours to give and the you
give it freely and irrevocably to the project.
How to Get Started:

1. Watch the “FusionPBX Pull Requests with Github” Youtube Video https://youtu.be/SPUe7S4Z6ms
2. If you have a good handle on PHP, Lua or SQL Development Might be the thing for you head over to the
Development Manual
3. Are you a FusionPBX power user and do you possible love to write? Check out the Documentation Guide or
the Testing Guide
4. Would you like to see FusionPBX in your native language and have the time to commit to staying on top of
translations for releases? Check out the translation section to learn how to use our translation server.
311
Note: Contributing Code or Documentation requires knowledge of Git, Github and how to create pull requests on
Github. This is not as bad as it sounds and if you are willing to learn we will help you through it.
312 Chapter 12. Contributing

CHAPTER 13
Documentation Guide
13.1 Documentation Guide
This page shows an nice overview of the reStructuredText syntax. This is not a comprehensive list of everything you
can do, but should be enough to get you up and running to contribute some really nice documentation. It is based on
resources found at Sphinx .
To get your own local documentation repository running, simply
13.1.1 Introduction
The reStructuredText (RST) syntax provides an easy-to-read, what-you-see-is-what-you-get plaintext markup syntax
and parser system. However, you need to be very precise and stick to some strict rules:
• like Python, RST syntax is sensitive to indentation !
• RST requires blank lines between paragraphs
This entire document is written with the RST syntax. In the right sidebar, you should find a link “Edit on Github”,
which will show each page in reStructuredText raw text format.
Contents
• Documentation Guide
– Introduction
– Getting Started
* Getting Git Right

* Setting up the Docs Locally
– Text Formatting
313
* Inline markup and special characters (e.g., bold, italic, verbatim)

* Headings
* Internal and External Links
* List and bullets
– What are directives
– Code and Literal blocks
* How to include simple code

* code-block directive
* Include code with the literalinclude directive
– Tables
* Simple tables
* Multicells tables, first method
* Multicells table, second method
* The tabularcolumns directive
* The csv-table directive
– The toctree directive
– Images and figures
* Include Images
* Include a Figure
– Boxes
* Colored boxes: note, seealso, todo and warnings

* Topic directive
* Sidebar directive
– Others
* Comments
* Substitutions
* glossary, centered, index, download and field list
* Footnote
* Citations
* More about aliases
* Intersphinx
* file-wide metadata
* metainformation
* contents directives
– Useful extensions
314 Chapter 13. Documentation Guide

* pngmath: Maths and Equations with LaTeX

* TODO extension
13.1.2 Getting Started
13.1.2.1 Getting Git Right
Learn Git in 15 Minutes Git Tutorial that will help you get started if you prefer. There is also awesome Git Tutorials
on the Atlassian Git site. Here is the link on installing Git if you don’t have it yet Git Install
13.1.2.2 Setting up the Docs Locally
One of the great things about Git and documentation is that all people who contribute are encouraged to setup their
own local copy of the docs for off-line editing. This by default will ensure that many backups of the documents exist
and there is never any concern about losing them.
Assuming you have Python already, install Sphinx locally:
$ pip install sphinx sphinx-autobuild
Clone the FusionPBX Github documentation repository:
$ cd /path/to/where_you_want_the_docs
$ git clone https://github.com/fusionpbx/fusionpbx-docs.git
$ cd fusionpbx-docs
Edit files or add new ones then build your changes:
$ make html
Open index.html with your web browser and check your changes:
fusionpbx-docs/build/html/index.html
Edit your files and rebuild until you like what you see, then commit your changes and push to the public repository.
Assuming the file you changed is called myfile.rst:
$ git add myfile.rst

$ git commit -m 'your commit message'
$ git push -u origin master
13.1.3 Text Formatting
13.1.3.1 Inline markup and special characters (e.g., bold, italic, verbatim)
There are a few special characters used to format text. The special character * is used to defined bold and italic text
as shown in the table below. The backquote character ` is another special character used to create links to internal or
external web pages as you will see in section Internal and External Links.
13.1. Documentation Guide 315

usage syntax HTML rendering

italic *italic* italic
bold **bold** bold
link `python <www.python.org>`__ python
verbatim ``*`` *
The double backquote is used to enter in verbatim mode, which can be used as the escaping character. There are some
restrictions about the * and `` syntax. They
• cannot not be nested,
• content may not start or end with whitespace: * text* is wrong,
• it must be separated from surrounding text by non-word characters like a space.
The use of backslash is a work around to second previous restrictions about whitespaces in the following case:
• this is a *longish* paragraph is correct and gives longish.
• this is a long*ish* paragraph is not interpreted as expected. You should use this is a
long\ *ish* paragraph to obtain longish paragraph
In Python docstrings it will be necessary to escape any backslash characters so that they actually reach reStructured-
Text. The simplest way to do this is to use raw strings by adding the letter r in front of the docstring.
Python string Typical result

r"""\*escape* \`with` "\\"""" *escape* `with` "\"
"""\\*escape* \\`with` "\\\\"""" *escape* `with` "\"
"""\*escape* \`with` "\\"""" escape with ""
13.1.3.2 Headings
In order to write a title, you can either underline it or under and overline it. The following examples are correct titles.
*****
Title
*****
subtitle
########
subsubtitle
**********************
and so on
Two rules:
• If under and overline are used, their length must be identical
• The length of the underline must be at least as long as the title itself
Normally, there are no heading levels assigned to certain characters as the structure is determined from the succession
of headings. However, it is better to stick to the same convention throughout a project. For instance:
• # with overline, for parts
• * with overline, for chapters
• =, for sections

• -, for subsections
• ^, for subsubsections
• “, for paragraphs
13.1.3.3 Internal and External Links
In Sphinx, you have 3 type of links:

1. External links (http-like)
2. Implicit links to title
3. Explicit links to user-defined label (e.g., to refer to external titles).
External links
If you want to create a link to a website, the syntax is
`<http://www.python.org/>`_
which appear as http://www.python.org/ . Note the underscore after the final single quote. Since the full name of the
link is not always simple or meaningful, you can specify a label (note the space between the label and link name):
`Python <http://www.python.org/>`_
The rendering is now: Python.
Note: If you have an underscore within the label/name, you got to escape it with a ‘’ character.
Implicit Links to Titles
All titles are considered as hyperlinks. A link to a title is just its name within quotes and a final underscore:
Ìnternal and External links`_
This syntax works only if the title and link are within the same RST file. If this is not the case, then you need to create
a label before the title and refer to this new link explicitly, as explained in Explicit Links section.
Explicit Links
You can create explicit links within your RST files. For instance, this document has a label at the top called
rst_tutorial, which is specified by typing:
.. _rst_tutorial:
You can refer to this label using two different methods. The first one is:
rst_tutorial_
The second method use the ref role as follows:

:ref:`rst_tutorial`
With the first method, the link appears as rst_tutorial, whereas the second method use the first title’s name found after
the link. Here, the second method would appear as Documentation Guide.
Note: Note that if you use the ref role, the final underscore is not required anymore.
13.1.3.4 List and bullets
The following code:
* This is a bulleted list.

* It has two items, the second
item uses two lines. (note the indentation)
1. This is a numbered list.

2. It has two items too.
#. This is a numbered list.

#. It has two items too.
gives:
• This is a bulleted list.
• It has two items, the second item uses two lines. (note the indentation)
Note: if two lists are separated by a blanck line only, then the two lists are not differentiated as you can see above.
13.1.4 What are directives
Sphinx and the RST syntax provides directives to include formatted text. As an example, let us consider the code-block
syntax. It allows to insert code (here HTML) within your document:
.. code-block:: html
:linenos:
<h1>code block example</h1>
Its rendering is:
1 <h1>code block example</h1>
Here, code-block is the name of the directive. html is an argument telling that the code is in HTML format, lineos is
an option telling to insert line number and finally after a blank line is the text to include.

Note that options are tabulated.
13.1.5 Code and Literal blocks
13.1.5.1 How to include simple code
This easiest way to insert literal code blocks is to end a paragraph with the special marker made of a double coulumn
::. Then, the literal block must be indented:
This is a simple example::
import math
print 'import done'
or:
This is a simple example:

::
import math
print 'import done'
gives:
This is a simple example:
import math
print 'import done'
13.1.5.2 code-block directive
By default the syntax of the language is Python, but you can specify the language using the code-block directive as
follows:
.. code-block:: html
:linenos:
<h1>code block example</h1>
produces
1 <h1>code block example</h1>
13.1.5.3 Include code with the literalinclude directive
Then, it is also possible to include the contents of a file as follows:
.. literalinclude:: filename
:linenos:
:language: python
:lines: 1, 3-5
:start-after: 3
:end-before: 5

13.1.6 Tables
There are several ways to write tables. Use standard reStructuredText tables as explained here. They work fine in
HTML output, however, there are some gotchas when using tables for LaTeX output.
The rendering of the table depends on the CSS/HTML style, not on sphinx itself.
13.1.6.1 Simple tables
Simple tables can be written as follows:
+---------+---------+-----------+
| 1 | 2 | 3 |
+---------+---------+-----------+
which gives:
1 2 3
Size of the cells can be adjusted as follows:
+---------------------+---------+---+
|1 | 2| 3 |
+---------------------+---------+---+
renders as follows:
1 2 3
This syntax is quite limited, especially for multi cells/columns.
13.1.6.2 Multicells tables, first method
A first method is the following syntax:
+------------+------------+-----------+
| Header 1 | Header 2 | Header 3 |
+============+============+===========+
| body row 1 | column 2 | column 3 |
+------------+------------+-----------+
| body row 2 | Cells may span columns.|
+------------+------------+-----------+
| body row 3 | Cells may | - Cells |
+------------+ span rows. | - contain |
| body row 4 | | - blocks. |
+------------+------------+-----------+
gives:

Header 1 Header 2 Header 3

body row 1 column 2 column 3
body row 2 Cells may span columns.
body row 3 Cells may span rows. • Cells
• contain
body row 4 • blocks.
13.1.6.3 Multicells table, second method
The previous syntax can be simplified:
===== ===== ======

Inputs Output
------------ ------
A B A or B
===== ===== ======
False False False
True False True
===== ===== ======
gives:
Inputs Output
A B A or B
False False False
True False True
Note: table and latex documents are not yet compatible in sphinx, and you should therefore precede them with the a
special directive (.. htmlonly::)
13.1.6.4 The tabularcolumns directive
The previous examples work fine in HTML output, however there are some gotchas when using tables in LaTeX: the
column width is hard to determine correctly automatically. For this reason, the following directive exists:
.. tabularcolumns:: column spec
This directive gives a âCœcolumn specâC for the next table occurring in the source file. It can have values like:
|l|l|l|
which means three left-adjusted (LaTeX syntax). By default, Sphinx uses a table layout with L for every column. This
code:
.. tabularcolumns:: |l|c|p{5cm}|
+--------------+---+-----------+
| simple text | 2 | 3 |
+--------------+---+-----------+
gives

title
simple text 2 3
13.1.6.5 The csv-table directive
Finally, a convenient way to create table is the usage of CSV-like syntax:
.. csv-table:: a title
:header: "name", "firstname", "age"
:widths: 20, 20, 10
"Smith", "John", 40
"Smith", "John, Junior", 20
that is rendered as follows:
Table 1: a title
name firstname age
Smith John 40
Smith John, Junior 20
13.1.7 The toctree directive
Sooner or later you will want to structure your project documentation by having several RST files. The toctree
directive allows you to insert other files within a RST file. The reason to use this directive is that RST does not have
facilities to interconnect several documents, or split documents into multiple output files. The toctree directive looks
like
.. toctree::
:maxdepth: 2
:numbered:
:titlesonly:
:glob:
:hidden:
intro.rst
chapter1.rst
chapter2.rst
It includes 3 RST files and shows a TOC that includes the title found in the RST documents.
Here are a few notes about the different options
• maxdepth is used to indicates the depth of the tree.
• numbered adds relevant section numbers.
• titlesonly adds only the main title of each document
• glob can be used to indicate that * and ? characters are used to indicate patterns.
• hidden hides the toctree. It can be used to include files that do not need to be shown (e.g. a bibliography).
The glob option works as follows:

.. toctree::
:glob:
intro*
recipe/*
*
Note also that the title that appear in the toctree are the file’s title. You may want to change this behaviour by changing
the toctree as follows:
.. toctree::
:glob:
Chapter1 description <chapter1>
So that the title of this section is more meaningful.
13.1.8 Images and figures
13.1.8.1 Include Images
Use:
.. image:: _static/images/logo.png
:width: 200px
:align: center
:height: 100px
:alt: alternate text
to put an image
13.1.8.2 Include a Figure
.. figure:: _static/images/logo.png
:width: 200px
:align: center
:height: 100px
:alt: alternate text
:figclass: align-center
figure are like images but with a caption
and whatever else youwish to add
.. code-block:: python
import image

gives
Fig. 1: figure are like images but with a caption

and whatever else youwish to add
import image
The option figclass is a CSS class that can be tuned for the final HTML rendering.
13.1.9 Boxes
13.1.9.1 Colored boxes: note, seealso, todo and warnings
There are simple directives like seealso that creates nice colored boxes:
See also:
This is a simple seealso note.
created using:
.. seealso:: This is a simple **seealso** note.
You have also the note directive:
Note: This is a note box.
with
.. note:: This is a **note** box.
and the warning directive:
Warning: note the space between the directive and the text
generated with:
.. warning:: note the space between the directive and the text
There is another todo directive but requires an extension. See Useful extensions
13.1.9.2 Topic directive
A Topic directive allows to write a title and a text together within a box similarly to the note directive.
This code:

.. topic:: Your Topic Title
Subsequent indented lines comprise

the body of the topic, and are
interpreted as body elements.
gives
Your Topic Title
Subsequent indented lines comprise the body of the topic, and are interpreted as body elements.
13.1.9.3 Sidebar directive
It is possible to create sidebar using the following code:
.. sidebar:: Sidebar Title

:subtitle: Optional Sidebar Subtitle
Subsequent indented lines comprise

the body of the sidebar, and are
interpreted as body elements.
Sidebar Title
Optional Sidebar Subtitle
Subsequent indented lines comprise the body of the sidebar, and are interpreted as body elements.
13.1.10 Others
13.1.10.1 Comments
Comments can be made by adding two dots at the beginning of a line as follows:
.. comments
13.1.10.2 Substitutions
Substitutions are defined as follows:
.. _Python: http://www.python.org/
and to refer to it, use the same syntax as for the internal links: just insert the alias in the text (e.g., Python_, which
appears as Python ).
A second method is as follows:

.. |longtext| replace:: this is a very very long text to include
and then insert |longtext| wherever required.
13.1.10.3 glossary, centered, index, download and field list
Field list
Whatever this is handy to create new field and the following text is indented
:Whatever: this is handy to create new field
glossary
.. glossary::
apical
at the top of the plant.
gives
apical at the top of the plant.
index
.. index::
download
:download:`download samplet.py <_downloads/sample.py>`
gives download sample.py
hlist directive
hlist can be use to set a list on several columns.

.. hlist::
.. hlist::
:columns: 3
* first item
* second item
* 3d item
* 4th item
* 5th item
• first item

• second item
• 3d item
• 4th item
• 5th item
13.1.10.4 Footnote
For footnotes, use [#name]_ to mark the footnote location, and add the footnote body at the bottom of the document
after a â CœFootnotesâC rubric heading, like so:
Some text that requires a footnote [#f1]_ .
.. rubric:: Footnotes
.. [#f1] Text of the first footnote.
You can also explicitly number the footnotes ([1]_) or use auto-numbered footnotes without names ([#]_). Here is
an example [#footnote1]_.
13.1.10.5 Citations
Citation references, like [CIT2002] may be defined at the bottom of the page:
.. [CIT2002] A citation
(as often used in journals).
and called as follows:
[CIT2002]_
13.1.10.6 More about aliases
Directives can be used within aliases:
.. |logo| image:: _static/images/logo.png

:width: 20pt
:height: 20pt
Using this image alias, you can insert it easily in the text |logo|, like this . This is especially useful when dealing
with complicated code. For instance, in order to include 2 images within a table do as follows:
+---------+---------+-----------+
| |logo| | |logo| | |longtext||
+---------+---------+-----------+
this is a longish text to include within a table and which is longer than the width of the column.
Note: Not easy to get exactly what you want though.

13.1.10.7 Intersphinx
When you create a project, Sphinx generates a file containing an index to all the possible links (title, classes, functions,
. . . ).
You can refer to those index only if Sphinx knowns where to find this index. THis is possible thanks to the intersphinx
option in your configuration file.
For instance, Python provides such a file, by default Sphinx knows about it. The following code can be found at the
end of a typical Sphinx configuration file. Complete it to your needds:
# Example configuration for intersphinx: refer to the Python standard library.

intersphinx_mapping = {'http://docs.python.org/': None, }
13.1.10.8 file-wide metadata
when using the following syntax:
:fieldname: some contents
some special keywords are recognised. For instance, orphan, nocomments, tocdepth.
An example of rendering is the toctree of top of this page.
orphan
Sometimes, you have an rst file, that is not included in any rst files (when using include for instance). Yet, there are
warnings. If you want to supprresse the warnings, include this code in the file:
:orphan:
There is also tocdepth and nocomments metadata. See Sphinx homepage.
13.1.10.9 metainformation
.. sectionauthor:: name <email>

Specifies the author of the current section.:
.. sectionauthor:: John Smith <js@python.org>
By default, this markup isnâ C™t reflected in the output in any way, but you can set the configuration value
show_authors to True to make them produce a paragraph in the output.
13.1.10.10 contents directives
.. contents::
.. contents:: a title for the contents

:depth: 2
• depth indicates the max section depth to be shown in the contents

13.1.11 Useful extensions
In the special file called conf.py, there is a variable called extensions. You can add extension in this variable. For
instance:
extensions = [-
'easydev.copybutton',
'sphinx.ext.autodoc',
'sphinx.ext.autosummary',
'sphinx.ext.coverage',
'sphinx.ext.graphviz',
'sphinx.ext.doctest',
'sphinx.ext.intersphinx',
'sphinx.ext.todo',
'sphinx.ext.coverage',
'sphinx.ext.pngmath',
'sphinx.ext.ifconfig',
'matplotlib.sphinxext.only_directives',
'matplotlib.sphinxext.plot_directive',
]
13.1.11.1 pngmath: Maths and Equations with LaTeX
The extension to be added is the pngmath from sphinx:
extensions.append('sphinx.ext.pngmath')
In order to include equations or simple Latex code in the text (e.g., 𝛼 ≤ 𝛽 ) use the following code:
:math:`\alpha > \beta`
Warning: The math markup can be used within RST files (to be parsed by Sphinx) but within your python’s
docstring, the slashes need to be escaped ! :math:`\alpha` should therefore be written :math:`\\alpha`
or put an “r” before the docstring
Note also, that you can easily include more complex mathematical expressions using the math directive:
.. math::
n_{\mathrm{offset}} = \sum_{k=0}^{N-1} s_k n_k
Here is another:
𝑁
∑︁−1
𝑛offset = 𝑠𝑘 𝑛𝑘
𝑘=0
It seems that there is no limitations to LaTeX usage:

𝑘−1
∏︁ 𝑁
∏︁−1
𝑠column
𝑘 = 𝑑𝑗 , 𝑠row
𝑘 = 𝑑𝑗 .
𝑗=0 𝑗=𝑘+1

13.1.11.2 TODO extension
Similarly to the note directive, one can include todo boxes but it requires the sphinx.ext.todo extension to be added in
the conf.py file by adding two lines of code:
extensions.append('sphinx.ext.todo')
todo_include_todos=True
.. todo:: a todo box
.. rubric:: Footnotes
.. [#footnote1] this is a footnote aimed at illustrating the footnote capability.
.. rubric:: Bibliography
.. [CIT2002] A citation
(as often used in journals).

CHAPTER 14
Other
14.1 Other
Other section is bits of info that needs indexed for the PDF to populate all sections depending how sections are
formatted.
14.1.1 XMPP Manager
XMPP Manager is an optional menu item. In order to have the option for XMPP Manager there are a few step to take
to enble XMPP.
XMPP Profile
• FusionPBX menu.
• Accounts -> XMPP manager.
• Click the plus on the right to create a profile.
Note: Google has since depricated xmpp service
In this example we will setup Google Talk and by creating a profile called gtalk.
331
Profile Name: gtalk

Username: your_user_account@gmail.com (use your account)
Password: use the correct password
Auto-Login: yes
XMPP Server: talk.google.com
Two approaches can be used for the next part.

Option 1.
Lets say my gmail number was 13051231234. This approach will send the inbound calls to the inbound routes with a
destination number that is the default extension number that is set.
Default extension: 13051231234

Advanced -> Context: public
Option 2.
332 Chapter 14. Other

On a single tenant system. This will send the call to extension 1001 in the default context.
Advanced -> Context: default
Option 3.
On a single tenant system. This will send the call to extension 1001 in the multi-tenant domain name.
Advanced -> Context: your.domain.com
Save the settings and restart the module. Restart the ‘XMPP’ module from Advanced -> Modules page. Go back to
Accounts -> XMPP if the status says ‘AUTHORIZED’ then you are ready to go.
Note If you are not getting AUTHORIZED you might need to goto the google account settings and choose “Allow
less secure apps: ON” under the Sign-in & security section.
Outbound Routes
For this example we will use 11 digit dialing.
Gateway: XMPP
Dialplan Expression: 11 digits
Description: Google Talk
Press Save
If your XMPP profile is named something other than gtalk edit the outbound route you just created. Bridge statement
should look like: dingaling/gtalk/+$1@voice.google.com replace gtalk with the profile name you chose and then save
it.
14.1.1.1 Enable XMPP
XMPP manager is used to configure client side XMPP profiles. It can be used as a client to register to make and
receive call with Google Talk or other XMPP servers.
GIT Manually add XMPP
After version 3.8 XMPP is optional. To add XMPP do the following
Goto command line
cd /tmp
git clone https://github.com/fusionpbx/fusionpbx-apps.git
cd fusionpbx-apps/
mv xmpp/ /var/www/fusionpbx/app/
14.1. Other 333


cd /var/www/fusionpbx/app
chown www-data:www-data -R xmpp/
Goto Fusionpbx GUI

Goto the GUI and click advanced > menu manager > edit icon > click “Restore Defaults” at top right
Then goto Advanced > Upgrade click Schema, Data Types, and Permission Defaults then click execute
Click status > sip status > Flush Memcache
Log out then back in
You should now have XMPP Manager under Accounts.
14.1.2 WebRTC
WebRTC app for FusionPBX is made by editing an existing FusionPBX app code and adding the code from the
“Master FreeSWITCH code example”. Also, keep in mind that you will need ssl certs working on the server.
Note: There are two “sets” of code in this app. One being an existing app from FusionPBX and the code example
from Master FreeSWITCH book in Chapter 8.
14.1.2.1 Prerequisites
• Working install of FusionPBX

• Working set of SSL certs (Not self signed) on said install of FusionPBX
• Working mod_verto setup.

• Patience
14.1.2.2 Install Steps
On your server
cd /usr/src
git clone https://github.com/fusionpbx/fusionpbx-apps
Move the directory 'webrtc' into your main FusionPBX directory
mv fusionpbx-apps/webrtc /var/www/fusionpbx/app
chown -R www-data:www-data /var/www/fusionpbx/app/webrtc
Log into the FusionPBX webpage

Advanced -> Upgrade
Menu Defaults and Permission Defaults.
Log out and back in.
14.1.3 Parking
Call “parking” transfers a current call to an available park extension, where the caller will listen to Music on Hold.
The extension that originally received the call is now free to accept other calls or direct another extension to join the
call that was parked.
For example: The receptionist receives a call, and the caller would like to speak to the engineering department. The
receptionist says “please hold while I transfer you,” and presses the PARK1 button. The call is sent to extension
5901 and the caller listens to music on hold. The receptionist is now free to make a call to her engineering staff,
or pages the engineering page group and says “Engineering you have a call on PARK1.”
The Engineer can press the flashing park button on his phone, and he will be connected to the caller, and the
park extension will be freed for another call.
Multiple park extensions can be created. Phones can be programmed with BLF functionality for parked exten-
sions, so the users can see if there is a call in that extension.
Below is an example of how to provision a Yealink SIP-T32G, which has 3 Line buttons.
14.1. Other 335


Index
A
apical, 326
C
contents (directive), 328
H
hlist (directive), 326
S
sectionauthor (directive), 328
337

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What is FusionPBX?

What is FusionPBX?

What are some features of FusionPBX?

What are some features of FusionPBX?

FusionPBX Documentation

Apr 03, 2020

11 Additional Information 265

13 Documentation Guide 313

1.1 Benefits of FusionPBX

1. Adding extra functionality to the incredibly robust FreeSWITCH VoIP Platform.

1.2 FusionPBX Features

Call Block Call Broadcast Call Flows Call Center

1.3 Our Ecosystem

2.1 Getting Started

2.1.1.2 Paid Training and technical support

Come join us! We look forward to meeting you!

2.1.2 Quick Install

6 Chapter 2. Getting Started

Welcome to the FusionPBX installation guide.

cd /usr/src/fusionpbx-install.sh/debian && ./install.sh

apt-get update && apt-get upgrade

Installation has completed.

Use a web browser to login.

2.1. Getting Started 7

(continued from previous page)

Official FusionPBX Training

8 Chapter 2. Getting Started

2.1.3 Let’s Encrypt

2.1.3.1 Dehydrated (Recommended)

To create a hostname or multiple hostname SSL certificate go to:

Then execute the script.

You should then see and follow the prompts.

Domain Name: domain.tld

After that, you should see the following output.

2.1. Getting Started 9

Cloning into 'dehydrated'...

nginx: the configuration file /etc/nginx/nginx.conf syntax is ok

To create a wildcard SSL certificate go to:

Then execute the script.

You should then see and follow the prompts:

Domain Name: *.domain.tld

Cloning into 'dns-01-manual'...

10 Chapter 2. Getting Started

(continued from previous page)

Add the following to the zone definition of domain.tld:

**Press enter to continue...**

+ Responding to challenge for domain.tld authorization...

**Press enter to continue...**

nginx: the configuration file /etc/nginx/nginx.conf syntax is ok

2.1. Getting Started 11

;; ANSWER SECTION: _acme-challenge.domain.tld. 1799 IN TXT “PY7ttk6no_5eG7WtAbO6qs5-NzA-

Setup for multiple domains on Let’s Encrypt

ssl_protocols TLSv1 TLSv1.1 TLSv1.2;

rewrite "^.*/provision/MN_Generic.cfg" /app/provision/index.php?mac=08000f000000&

12 Chapter 2. Getting Started

(continued from previous page)

rewrite "^.*/provision/([A-Fa-f0-9]{12})-phone.cfg$" /app/provision/?mac=$1;

rewrite "^.*/provision/([0-9]{1,11})_Phonebook.xml$" "/app/provision/?ext=$1&file={

# Disable viewing .htaccess & .htpassword & .db

Create a file to contain config for additional domains

Press enter to continue...

Press enter to continue...