BlueBox

BlueBox is a web based php configuration and management GUI for FreeSWITCH and Asterisk switching libraries. It supports multi-tenancy, skinning, and is completely open-source. It can be used with database and file replication to scale up to thousands of registered devices and simultaneous phone calls. It can operate in the cloud or on the premise. It originally developed from FreePBXv3.

I found the following two blogs on VOIP PBX useful:
 * freeSWITCH --- which covers Asterisk+FreePBX, FreeSWITCH+BlueBox and FreeSWITCH+FusionPBX
 * FreeSWITCH Cookbook

Prerequisites
I installed the following prerequisites:
 * -9999
 * -2.2.22-r1
 * -5.3.13
 * -5.1.62-r1

FreeSWITCH
Please first install FreeSWITCH following the FreeSWITCH Wiki

There are many USE options for FreeSwitch. Not all of them compile for me. [TODO] -- Add list here of which work and which don't compile.

After installing FreeSWITCH, there is a post-installation message from portage that a backup of the default configuration files has been saved to:

But a "diff -r /etc/freeswitch /usr/share/doc/freeswitch-9999/conf" shows differences. So I made my own copy:

Also, portage seems to install the freeswitch configuration directory twice: A "diff" for each file/folder in /etc/freeswitch/* and /etc/freeswitch/freeswitch/* shows no differences. Is this a mistake?
 * once into
 * and a second time into

Double check FreeSWITCH permissions: -- I think during BlueBox installation I may have done a "chown -R apache.freeswitch /opt/freeswitch". If apache needs read access to during BlueBox installation, the ownership/permissions need to be reset as follows or you cannot run freeswitch as user "freeswitch":

Also, the script tries to change the pid file from the default for FreeSwitch from  to. FreeSwitch insists on writing its pid to, but the Gentoo init script looks for the pid in. So my solution is to change the init script:

Web Server
Install a web server such as
 * Apache
 * nginx

PHP
Install PHP.

Database
Install a database backend such as:
 * MySQL
 * PostgreSQL

Database Setup
Setup the MySQL database:

If you make a mistake and need to start over:

Installing Bluebox
Adapted from Bluebox FreeSwitch install guide (CentOS v5.x, Freeswitch v1.0.x, Bluebox).

Install BlueBox git source into top level root of web server:

When the install asks you for the web user name change default ("apache") to "freeswitch" (without quotes). Everything else can remain at default (just repeatedly press when it asks).

Increase the memory_limit for PHP:

Restart apache:

is a symlink to. You will need to change the ownership on the configuration files to allow apache to make changes before you run the BlueBox configuration GUI:

Browse to "http://your.web.server/bluebox/" and run the installer. I went with the defaults but changed the password and email address for the administrator. I also unchecked the option for installing sample data.

I got a warning about conflicting files:

I just continued.

After configuring BlueBox, my default configuration was as follows:

You probably will want to change the password for the user "bluebox".

Next edit and disable the installer:

Starting FreeSWITCH
There appears to be no man installation for FreeSWITCH. So use --help to get freeswitch command line options:

I got an error when starting FreeSWITCH:

So I just added ulimit -s 240 to the start routine of.

Some useful fs_cli (FreeSWITCH command line interface) for debugging, tracing: Package Manager A few modules have prerequisite modules so if you get an error, install the prerequisite. A few pairs of modules conflict, e.g.: I chose "Freeswitch", "Endpoint Manager" and "Media File" and installed every remaining module except those for a call center.
 * "Freeswitch" and "Asterisk"
 * "Endpoint Manager" and "Provisioner"
 * "Media File" and "Media Manager"

Bluebox came pre-installed with three sip interfaces "Connectivity" -> "SIP Interface" Authenticated SIP	 Auto Detect	5060	Required   Edit|Delete Authenticated SIP - NAT Auto Detect	5070	Required   Edit|Delete Unauthenticated SIP	 Auto Detect	5080	None	   Edit|Delete

Use the FreeSWITCH CLI to see your interfaces: freeswitch@myhost> sofia status Name         Type                                       Data      State =================================================================================================          sipinterface_2       profile            sip:mod_sofia@192.168.1.40:5070      RUNNING (0) voicemail_1        alias                             sipinterface_2      ALIASED sipinterface_1      profile            sip:mod_sofia@192.168.1.40:5060      RUNNING (0) sipinterface_3      profile            sip:mod_sofia@192.168.1.40:5080      RUNNING (0) 192.168.1.40        alias                             sipinterface_1      ALIASED ================================================================================================= 3 profiles 2 aliases

Now configure users, devices, voicemail boxes, endpoints and assign numbers. It facilitates configuring your sip device if you first configure and then
 * the Users ("Organization" -> "User Manager")
 * their Voicemail Boxes ("Applications" -> "Voicemail Boxes")
 * the Endpoints ("Applications" -> "Endpoints")
 * the sip phones ("Applications" -> "Devices")

The passwords for the users must be alphanumeric but the passwords for the sip devices and voicemail boxes need not be so. To keep things simple I used the same identifying string for device names, sip accounts, extensions. For example, Device Name "101" associated with user "John Doe" used sip account "101" and was assigned extension "101" and unanswered calls were transferred to the voicemail box for 101 (extension 201) All used the same password except for the the required alphanumeric password for the User Account for John Doe.

If you decide to have a default Multitenant system, then your sip registrations will use "user@domain" instead of just "user" for the User Account. Make sure that your user:password (or user@domain:password) for each Device in bluebox matches what you have configured for each sip phone.

First try to get sip registration working before attempting secure sip (sips) registration or encrypted media (srtp). FreeSWITCH also has support for end-to-end encryption using zrtp (see FreeSWITCH Wiki: ZRTP). But you must use a sip phone which supports zrtp such as the softphone Zfone. Unfortunately, my Snom phones don't support zrtp and Snom has no plans to do so.

Note that bluebox adds your sip devices to. So check and verify that your users have been added there.

Verify that your sip phones have registered with FreeSWITCH: sofia status profile sipinterface_1 ================================================================================================= Name                   sipinterface_1 Domain Name            N/A Auto-NAT               false DBName Pres Hosts Dialplan               XML Context                multitenant_routing_context Challenge Realm        auto_to RTP-IP                 192.168.1.40 SIP-IP                 192.168.1.40 URL                    sip:mod_sofia@192.168.1.40:5060 BIND-URL               sip:mod_sofia@192.168.1.40:5060 HOLD-MUSIC             N/A OUTBOUND-PROXY         N/A CODECS IN              G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT             G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT              101 DTMF-MODE              rfc2833 CNG                    13 SESSION-TO             0 MAX-DIALOG             0 NOMEDIA                false LATE-NEG               false PROXY-MEDIA            false AGGRESSIVENAT          false STUN-ENABLED           true STUN-AUTO-DISABLE      false CALLS-IN               2 FAILED-CALLS-IN        1 CALLS-OUT              1 FAILED-CALLS-OUT       1 REGISTRATIONS          4

Show registrations for sipinterface: sofia status profile sipinterface_1 reg <... excerpt ..> Call-ID:       3c26701f3482-2ou8k0j6yqug User:          101@192.168.1.40 Contact:       "John Doe" < sip:101@192.168.1.101:2048 > Agent:         snom360/8.4.32 Status:        Registered(UDP)(unknown) EXP(2012-05-18 20:26:09) EXPSECS(2869) Host:          pbx IP:            192.168.1.101 Port:          2048 Auth-User:     101 Auth-Realm:    pbx.mydomain.com MWI-Account:   101@voicemail_1

Now that our phones have registered with FreeSWITCH, check that you have a dial tone and can dial an internal extension.

IPKall
If you want to test SIP from outside your network, apply for a free DID from IPKall (Washington State). Just select an area code and choose a "SIP phone number" which can be any alphanumeric string, eg. "IPKall". Enter the "SIP Proxy" for your FreeSWITCH server, eg. "pbx.mydomain.com:5080". Make sure you specify a SIP port of 5080 instead of the default 5060. BlueBox uses different SIP ports to listen on for each different SIP interface, e.g.:
 * Authenticated SIP	  Auto Detect	5060
 * Authenticated SIP - NAT  Auto Detect	5070
 * Unauthenticated SIP	  Auto Detect	5080

Then go to "Routing -> Number Manager" and add your new number:
 * Manage -> Number: "IPKall"
 * Manage -> Type: Internal
 * Device (Pick a destination such as Device "101")
 * Contexts -> Inbound Routes
 * Number Pools -> Device

Now add a trunk for IPKall "Connectivity -> Trunk Manager":
 * Trunk Name: "IPKall"
 * Trunk Type: "SIP Interface"
 * Server: "voiper.ipkall.com"
 * Bind To Interface: "Unauthenticated SIP"
 * Made from these Contexts: "Inbound Routes"
 * Caller ID Name: "ipkall"
 * Caller ID Number: 

Now try phoning the IPKall DID phone number emailed to you. I got a busy signal, so I had to edit "Connectivity -> SIP Interfaces -> Unauthenticated SIP" and change "Inbound Calls -> Default Incoming Context" from "AUTO(Multitenant)" to "Inbound Routes" since I opted not to have a multitenant system.

NAT Traversal For SIP
See NAT Traversal RTP SIP

Configuring a Dialplan (TODO)
Now we need to configure various components for a dialplan. BlueBox's modules make configuring a dialplan rather straight forward. A few things from FreePBX seem to be missing: or to use text-to-speach (flite, cepstral) for your prompts It should be possible to do something similar with FS+BB. The configuration files created by BB begin with "bluebox_", can one though edit other configuration files and have them included in the BB dialplan? without getting clobbered by BB?
 * A simple way to call an extension to record an IVR prompt for use with an AutoAttendant
 * BB does however allow you to upload a pre-recorded media file
 * No iax or dahdi (freetdm) modules -- only sip
 * Early versions of FreePBX did not have a dahdi configuration utility for managing chan_dahdi_additional.conf, however you could still manually edit the chan_dahdi_custom.conf.

Auto Attendant

 * AutoAttendant

Time Based Routes

 * Time Based Routes

Ring Groups

 * Ring Groups

Conferences

 * Conferences

Feature Codes

 * Feature Codes such as CheckingVoicemail, etc.

Trunks

 * Trunks: DAHDI (PSTN), SIP, IAX

Saving your BlueBox Configuration
Your BlueBox configuration files are stored in ->

So make a copy:

Now make a symlink:

If you also installed FusionPBX, then do similarly:

Before starting freeswitch or using the GUI just remember to copy either to  or change the symlink:

Authenticated SIP - NAT Auto Detect	5070	Required   Edit|Delete > /dev/null