Building an SIP PBX with sipXPBX
DONE SipX Start Here
DONE sipX Stable Download Area
for Fedora and CentOS
Overview, Deployment & Architecture
install sipX 3.6 on Fedora Core 5
install sipX 3.4 on Fedora Core 5
Forum : General, FAQs, How to
SIPfoundry sipX - The Open Source SIP PBX for Linux
Configure DHCP and DNS Servers
Admin and User Manuals (requires free registration)
As of October 2006, sipXPBX this
is the most serious alternative to Asterisk. Unlike Asterisk, and as the name
implies, sipXPBX was written from the start to run as an SIP server, while Asterisk
was started years ago when things were as clear and H.323 seemed to be on its
way to become the standard in VoIP. For more information, read How
to Compare sipX ECS with the Asterisk PBX (sipX vs. Asterisk). Note that
Asterisk and sipXPBX can
be used together.
sipXPBX is built from the following parts:
- sipXproxy for call routing
- sipXregistry for registry/redirect
modular server for handling SIP event subscriptions
as a media server with auto-attendant and voice mail applications
for configuration and phone provisioning (written in Java)
- and PostGreSQL as the database
Here's a list
As of Oct 2006, the site is a bit of a mess: part of it is located on www.sipfoundry.org,
while the other part is located on http://sipx-wiki.calivia.com,
and installation instructions are located in different documents. Why is part of the sipX stuff located on http://www.calivia.com
? "Calivia is actively participating in the development of the Open Source IP PBX software sipX from SIPfoundry.
We currently maintain and host the sipX documentation Wiki.
The Wiki started as our own documentation site as we began working with
sipX on Gentoo Linux. After only two month of public operation it
already hosted a wealth of information regarding SIP technology in
general and the sipX software in particular."
Setting up Fedora 5
Setting up sipXPBX
- Download sipxPBX 3.6 Beta
wget -P /etc/yum.repos.d http://www.sipfoundry.org/pub/sipX/sipx-fc.repo
install --enablerepo=sipx-testing sipxpbx
(Until 3.6 is released, you'll
need to add --enablerepo=sipx-testing to every yum command.)
- Create a valid SSL certificate
"CA Common Name" is anything BUT NOT the DNS name of your server,
"SIP domain name" is the domain name of your installation, and
"Full DNS name for the server" is the fully qualified hostname
of your sipX server
- /sbin/service sipxpbx start OR /etc/init.d/sipxpbx start OR /etc/rc.d/init.d/sipxpbx
start. You'll see errors linked to PostGreSQL, but they'll go away the second
time you run sipX
- Aim your browser at the server http://sipx.your.domain
. First, you are asked to define a password for the superadmin user. Once
entered successfully, you should then login as user superadmin, and proceed
with creating users, phones, and trunks.
Managing the server
Add a group: User > User Groups > Add Group : PhoneUsers
Select group to view/change permissions
Add a couple of Users: Users > Users > Add User. Note that Groups doesn't
come with a combobox: you must type the group name(s) yourself
GAVE UP: Can't register phone, couldn't restart service, logging didn't work
- Create a user: name (+ extension as an alias), PIN to check voicemail
and logon to the configuration server as user; The SIP password
is auto-generated by the system to assure maximum security. Aliases are case-sensitive.
You can set up zero, one, or more aliases for a user using a comma separated list.
- Users can and should be members of one or several groups. Groups not
only allow you to logically group users together, but groups are also
used to define permissions. Make sure you set permissions for the user, either
on a per user basis by clicking on the Permissions link or by setting permissions
for the group the user is a member of.
The distinction between general permissions and call permissions
is not always perfectly intuitive, but, generally, call permissions influence call routing privileges (i.e.
access to dial plan) where general permissions determine access to
- After you've created users and phones, you need to assign users to their phones.
Click on Add Line. You can search for a specific user or just click Search with empty fields, which will list all users. Select a user and press Select. You have now attached that user to the device. Dependent on the phone model, several lines can be attached to a device.
Adding an external phone line
(needed? looks like link to remote PBX)
After adding at least one phone to your system, navigate to the phone you'd like to add the line to, and click on the "Lines" link on the left-hand side of the page.
Click on the "Add External Line" link and fill in the registration information for this line.
Adding an SIP Trunk
No information at http://sipx-wiki.calivia.com/index.php/How_to_configure_SIP_Trunks
Dial plan & routing
How to configure Dial Plans
How to Localize the Dial Plan
How to configure User Call Forwarding
How to configure Caller ID
How to configure Domain Aliases
HowTo Configure the sipX Voicemail Service
HowTo record custom voice prompts
HowTo configure the sipX Auto Attendant
Checking processes and configuration
Resetting the superadmin password
/usr/bin/sipxconfig.sh --database reset-superadmin
Changing user passwords
The SIP password required to authenticate with sipX is auto-generated by
sipX. It can be viewed / changed by pressing Show Advanced Settings.