Table of Contents
This quick start guide covers basic steps from installing sipXcom to placing internal and external calls. Whether you are installing thousands of phones or just setting up a demo system, sipXcom graphical user interface makes the process straightforward and easy.
Create DNS records for your domain
Create ISO DVD and boot from ISO DVD
Enter network settings
Enter host and domain settings
Update sipXcom with maintenance releases
Select server core, telephony and device services
Configure soft and or hard phones
Add SIP Trunking for incoming and outgoing telephone calls
DNS Records for Your Domain
sipXcom requires correct DNS settings to work. It can automatically configure its own DNS server or tell you what the settings need to be. To only test the admin UI, you don't need DNS setup and can use the IP address, but it is still good to at least have the A record for the host set.
A DNS Domain that is equivalent to the SIP domain
A-Record (host record) for the server
SRV records for the SIP communications (port 5060 tcp & udp).
SRV record for the resource record (port 5070 tcp)
SRV record for XMPP client connections (port 5222 tcp)
SRV record for XMPP server connections (port 5269 tcp)
SRV record for XMPP client connections to XMPP conference (port 5222)
SRV record for XMPP servers connections to XMPP conference (port 5222)
Please follow guidance in link below:
Obtain, Burn and Boot Installation DVD
Download a stable release or development ISO from here, burn the image on a physical DVD. The ISO contains CentOS Linux operating system and sipXcom and all related compenents.
Insert the DVD into your Intel or AMD server and power it up.
- Press enter on the boot screen to begin sipXcom installation
Select the language, keyboard and time zone to be used during the installation.
Set a root password for sipXcom server.
Click Next on Boot Loader screen.
Don't forget to take your DVD out of the drive or eject the virtual drive...
Initial sipXcom Configuration
Login to the system as root with the password you provided earlier and continue to configure sipXcom starting with Network Settings. Enter "n" response to "Would you like to configure your system's network settings?" and continue entering the rest of the items.
Set superadmin password
Using a computer with network connectivity to the newly installed server, launch a Web browser and go to the URL or IP address displayed by the setup wizard (just the hostname of your server).
- The first time you log in Configuration Server Web UI you have to set superadmin's password (enter a strong (hard-to-guess) password that you will not forget)
Log in Configuration Server Web UI
Use superadmin for User ID and password set in step before for logging in Web UI.
Update DNS external servers
Insert one or more external DNS servers that can resolve external names (System Menu -> DNS).
Update sipXcom with maintenance releases
Logout of the Admin Portal. Then sign in to sipXcom server as root.
Guidance on upgrading can be found at:
Configure Servers – Core Services
Log into Admin Portal. Select all the services checked and Apply. You can mouse over the services for a description of each service.
An easy approach is to enable DHCP services and place the phones on a dedicated network segment separate from installed data devices (and other DHCP services).
Configure Servers – Telephony Services
Select all the services checked and Apply. You can mouse over the services for a description of each service.
Configuring Servers - Instant Messaging
Select all and Apply.
Configure Servers – Device Provisioning
Select options checked and Apply. You can mouse over the services for a description of each service.
Click Users -> Users and then "Add New User"
User ID is automatically inserted but may be changed. At a minimum, check Enabled and enter Last Name, First Name, for user portal Password, and password for Voicemail. SIP password is automatically generated but may be changed. Check Apply and OK.
In either case, if a phone is not configured by sipXcom then it does NOT need a phone entry (as in the example of LINPHONE below)
An open source SIP soft phone, LINPHONE for Windows, MAC OSX and Linux can be downloaded from:
Configure your account and click Apply. Your LINPHONE will register on sipXcom and you are ready to make calls between extensions.
Username: must be a User ID already set up on sipXcom
Password: SIP Password for the above Username from sipXcom Advanced Settings on User Identification screen
Domain: Fully Qualified Domain Name (host plus domain) or use Domain name if you have SIP SRV records in the DNS zone.
Use Dialer to Make Calls
At the bottom of the screen, please see "Registration on sip:sip.nycsip.com successful."
Dialer reflects call in progress.
Hard Phone Configuration
- Navigate to Devices > Phones
ADD New Phone
sipXcom will automatically configure the many brands and models of hard phones shown in the drop down list shown on the next page. Information on phones can be found in the Wiki at http://wiki.sipxcom.org/display/sipXcom/Hardphones
Insert the serial number of the phone which is the MAC address. Also select the most current firmware version available on sipXcom for this model of phone.
Add New Lines
- After creating the phone navigate to User Lines tab and Add New Line
Perform the same steps for the second user - add new phone and assign user 201.
After creating the phones and assigning lines you have to send profiles to phones for the settings to become effective in the phones. In the phone main page select the phones and click Send Profiles button. Monitor status of action in Diagnostics > Job Status page.
- After sending profiles the phones will reboot and you can place a call from extension 200 to 201.
SIP Trunking is one way to provide the capability to connect to the Public Switched Telephone Network (PSTN). This will allow you to make a call to someone on the PSTN and to receive calls from others connected to the PSTN.
Setup Gateway Configuration
Contact an Interoperable ITSP provider to get account information and ITSP server information from list at:
Select Devices - Gateways - Add New Gateway - SIP Trunk
Insert Name of ITSP, use built in SIP Trunk SBC, use provider template and select from drop down, and FDQN of ITSP server. Click Apply and OK.
Set up Caller ID
Enter caller id and name. Click Apply and OK.
Select and enable Dial Plan.
Click on dial plan to be used with gateway.
Enable the dial plan.
Select Long Distance dial plan and then Apply and OK.
Set up ITSP Account
Enter ITSP Username and Authentication Username obtained from ITSP. Usually these are the same. Enter the ITSP password. Enter IPSP FQDN. Click Apply and Enter.
Use aliases on User Identification to forward incoming calls. Insert the 10 digit telephone number for incoming calls provided by the ITSP.
Place a telephone call
You are now ready to Send and Receive Calls from the Public Switch Telephone Network (PSTN).