BlueBox
Redwood 19:13, 19 May 2012 (UTC)
Contents |
OVERVIEW
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.
PREREQUISITES
I installed the following prerequisites:
- net-misc/freeswitch-9999
- www-servers/apache-2.2.22-r1
- dev-lang/php-5.3.13
- dev-db/mysql-5.1.62-r1
FreeSWITCH
Please first install FreeSWITCH (FS) following the Gentoo Wiki
There are many USE options for FS. Not all of them compile for me.
[TODO] -- Add list here of which work and which don't compile.
After installing FS, there is a post-installation message from portage that a backup of the default configuration files has been saved to: /usr/share/doc/freeswitch-9999/conf
but a "diff -r /etc/freeswitch /usr/share/doc/freeswitch-9999/conf" shows differences
so I made my own copy:
root # </code><code>ln -s /etc/freeswitch.orig /opt/freeswitch/conf.orig</code>
Also, portage seems to install the freeswitch configuration directory twice:
* once into /etc/freeswitch * and a second time into /etc/freeswitch/freeswitch
A "diff" for each file/folder in /etc/freeswitch/* and /etc/freeswitch/freeswitch/*
shows no differences. Is this a mistake?
DOUBLE CHECK FS PERMISSIONS: -- I think during BlueBox installation
I may have done a "chown -R apache.freeswitch /opt/freeswitch"
If apache needs read access to "/opt/freeswitch" during BlueBox installation,
the ownership/permissions need to be reset as follows
or you cannot run freeswitch as user "freeswitch"
chown -R root:freeswitch "/etc/freeswitch"
chmod -R u=rwX,g=rX,o= "/etc/freeswitch"
# # prefix
chown -R root:freeswitch "/opt/freeswitch"
chmod -R u=rwX,g=rX,o= "/opt/freeswitch"
# allow read access for things like building external modules
chmod -R u=rwx,g=rx,o=rx "/opt/freeswitch/"{lib*,bin,include}
chmod u=rwx,g=rx,o=rx "/opt/freeswitch"
# # directories owned by the fs user
for x in db run log cores storage recordings; do
chown -R freeswitch:freeswitch "/opt/freeswitch/${x}"
done
ALSO, the /etc/init.d/freeswitch script tries to change the pid file from the default for FS
"/opt/freeswitch/run/freeswitch.pid" to "/var/run/freeswitch.pid"
FS insists on writing its pid to /opt/freeswitch/run/freeswitch.pid
but the Gentoo init script looks for the pid in /var/run/freeswitch.pid
So my solution is to change the init script:
start() {
local OPTS
[ -n "${FREESWITCH_USER}" ] && \
OPTS="${OPTS} -u ${FREESWITCH_USER}"
[ -n "${FREESWITCH_GROUP}" ] && \
OPTS="${OPTS} -g ${FREESWITCH_GROUP}"
[ -n "${FREESWITCH_OPTS}" ] && \
OPTS="${OPTS} ${FREESWITCH_OPTS}"
ebegin "Starting Freeswitch"
ulimit -s 240
start-stop-daemon --start --quiet --exec ${FREESWITCH_BIN} \
--pidfile /opt/freeswitch/run/freeswitch.pid -- -nc ${OPTS}
eend $?
}
Then install:
WEB SERVER
INSTALL a web server such as www-servers/apache (see the Gentoo Wiki)
PHP
INSTALL dev-lang/php (see the Gentoo Wiki)
DATABASE
INSTALL a database backend such as:
- dev-db/mysql (see the Gentoo Wiki)
- dev-db/postgresql-server (see the Gentoo Wiki)
- dev-db/sqlite (see the Gentoo Wiki)
DATABASE SETUP
mysql -u root -p mysql> CREATE USER 'bluebox'@'localhost' IDENTIFIED BY 'bluebox'; mysql> GRANT ALL PRIVILEGES ON bluebox.* TO 'bluebox'@'localhost'; mysql> FLUSH PRIVILEGES; mysql> SELECT user,host,password FROM mysql.user; mysql> SHOW GRANTS FOR bluebox@localhost; mysql> \q
If you make a mistake and need to start over:
mysql -u root -p mysql> REVOKE ALL PRIVILEGES, GRANT OPTION ON bluebox.* FROM 'bluebox'@'localhost'; mysql> DROP DATABASE bluebox; mysql> \q
INSTALLING BLUEBOX (BB)
Adapted from http://www.powerpbx.org/content/centos-freeswitch-bluebox-v1
Install bluebox git source into top level root of web server:
cd /var/www/localhost/htdocs<br /> git clone git://source.2600hz.org/bluebox.git<br /> chown -R freeswitch:freeswitch /var/www/localhost/htdocs/bluebox<br /> cd /var/www/localhost/htdocs/bluebox<br /> ./preinstall.sh<br />
NOTE: 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 ENTER when it asks)
Increase the memory_limit for php
Restart apache
/opt/freeswitch/conf/ is a symlink to /etc/freeswitch
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:
Conflicting Files<br /> /opt/freeswitch/conf/directory/default.xml<br /> /opt/freeswitch/conf/autoload_configs/conference.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/ivr.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/acl.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/xml_cdr.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/callcenter.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/distributor.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/directory.conf.xml<br /> /opt/freeswitch/conf/autoload_configs/cdr_csv.conf.xml<br /> /opt/freeswitch/conf/sip_profiles/external.xml<br /> /opt/freeswitch/conf/sip_profiles/internal-ipv6.xml<br /> /opt/freeswitch/conf/sip_profiles/internal.xml<br /> Conflicting configuration files will be permanently erased if you continue!<br />
I just continued.
After configuring bluebox, my default configuration was
cat /var/www/localhost/htdocs/bluebox/config/database.php
'type' => 'mysql',
'user' => 'bluebox',
'pass' => 'bluebox',
'host' => '127.0.0.1',
'port' => '3306',
'socket' => FALSE,
'database' => 'bluebox'
You probably will want to change the password for user bluebox.
Next edit /var/www/localhost/htdocs/bluebox/bluebox/config/config.php
and disable the installer:
config['installer_enabled'] = FALSE;
STARTING FreeSWTICH
There appears to be no man installation for freeswitch
So use --help to get freeswitch command line options:
/opt/freeswitch/bin/freeswitch --help
these are the optional arguments you can pass to freeswitch
-nf -- no forking
-u [user] -- specify user to switch to
-g [group] -- specify group to switch to
-help -- this message
-version -- print the version and exit
-waste -- allow memory waste
-core -- dump cores
-rp -- enable high(realtime) priority settings
-lp -- enable low priority settings
-np -- enable normal priority settings (system defaults)
-vg -- run under valgrind
-nosql -- disable internal sql scoreboard
-heavy-timer -- Heavy Timer, possibly more accurate but at a cost
-nonat -- disable auto nat detection
-nonatmap -- disable auto nat port mapping
-nocal -- disable clock calibration
-nort -- disable clock clock_realtime
-stop -- stop freeswitch
-nc -- do not output to a console and background
-ncwait -- do not output to a console and background but wait until the system is ready before exiting (implies -nc)
-c -- output to a console and stay in the foreground
-conf [confdir] -- specify an alternate config dir
-log [logdir] -- specify an alternate log dir
-run [rundir] -- specify an alternate run dir
-db [dbdir] -- specify an alternate db dir
-mod [moddir] -- specify an alternate mod dir
-htdocs [htdocsdir] -- specify an alternate htdocs dir
-scripts [scriptsdir] -- specify an alternate scripts dir
I got an error when starting FS:
/opt/freeswitch/bin/freeswitch -c -rp -u freeswitch Error: stacksize -1 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... Cannot open pid file /opt/freeswitch/run/freeswitch.pid.
So I just added "ulimit -s 240" to the start routine of /etc/init.d/freeswitch
Some useful fs_cli (freeswitch command line interface) for debugging, tracing:
sofia status
sofia status profile sipinterface_1
sofia status profile sipinterface_1 reg
sofia loglevel all 9
sofia global siptrace on
console loglevel debug
eval${domain}
expand sofia contact <sip user account>
CONFIGURING FS WITH BLUEBOX
BROWSE to "http://your.web.server/bluebox/"
INSTALL additional modules:
System->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.:
- "Freeswitch" and "Asterisk"
- "Endpoint Manager" and "Provisioner"
- "Media File" and "Media Manager"
I chose "Freeswitch", "Endpoint Manager" and "Media File"
and installed every remaining module except those for a call center.
Bluebox came pre-installed with 3 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
- the Users ("Organization" -> "User Manager")
- their Voicemail Boxes ("Applications" -> "Voicemail Boxes")
- the Endpoints ("Applications" -> "Endpoints")
and then
- 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 the FreeSWITCH Wiki) but you must use a sip phone which supports zrtp such as 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 /etc/freeswitch/directory/default.xml
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 FreeSWITH 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: <DID number emailed to you by IPKall>
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
Next caveat is that I use NAT and need to forward the
following udp ports from my router to my FS server:
- 5060 (standard SIP)
- 5070
- 5080
as well as the port range for rtp (10000 to 20000)
In addition I use shorewall for the firewall on my FS server,
so I need the following in my /etc/shorewall/rules:
# ASTERISK and FreeSWITCH ACCEPT loc $FW tcp 5038 # AMP -- Asterisk Manager Protocol ACCEPT all $FW udp 5036 # iax ACCEPT all $FW udp 4569 # iax2 # SIP ACCEPT all $FW udp 5060 # sip ACCEPT all $FW tcp 5060 # sip Some SIP servers need tcp as well ACCEPT all $FW udp 5061 # sips ACCEPT all $FW tcp 5061 # sips ACCEPT all $FW udp 5070 # FreeSWITCH sip udp ACCEPT all $FW tcp 5070 # FreeSWITCH sip tcp ACCEPT all $FW udp 5080 # FreeSWITCH sip udp ACCEPT all $FW tcp 5080 # FreeSWITCH sip tcp ACCEPT all $FW udp 10000:20000 # rtp #ACCEPT all $FW udp 2727 # MGCP media gateway control protocol #H323 tcp 522, 1503, 1720, 1731 and 8080
I also disabled connection tracking for sip in /etc/shorewall/start
modprobe -r nf_nat_sip &> /dev/null
modprobe -r nf_conntrack_sip &> /dev/null
#
modprobe -r nf_nat_h323 &> /dev/null
modprobe -r nf_conntrack_h323 &> /dev/null
And I also added the above ports to /etc/shorewall/notrack:
NOTRACK loc 192.168.1.0/24 udp 5060 # sip
NOTRACK loc 192.168.1.0/24 tcp 5060 # sip
NOTRACK $FW 192.168.1.0/24 udp 5060 # sip
NOTRACK $FW 192.168.1.0/24 tcp 5060 # sip
and so on for ports 5061, 5070, 5080, 10000:20000
But (not unexpectedly) after calling the DID and picking up my extension, there was no audio
on either my cell phone or sip phone. So next I edited the "Unauthenticated SIP" interface
and changed the "Default Interface" to "Authenticated SIP -- NAT" and checked the following:
- Server is behind NAT?: ON
- NAT detection mechanism: "Detect IP via uPnP"
- SIP Ping Registered Devices: ON
- Aggressive NAT Detection: ON
- Use Network IP & Port for RTP: OFF
Now I had audio on my cell phone but still no audio on my sip phone.
[TODO] FIX NAT traversal for SIP on my FS server.
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:
- 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
or to use text-to-speach (flite, cepstral) for your prompts
- 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.
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?
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
==== ====
