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.
sipXcom scales to tens of thousands of users and has high availability features required for large installations. sipXcom has extensive features necessary to replace legacy telephone systems and includes significant unified communications capabilities.
For large and complex installations, sipXcom provides high reliability, rich functionality, significant reductions in capital expenses (CAPEX) and significant reductions in operating expenses (OPEX). CAPEX is reduced by cost-effective features such as sipXcom running on standard Intel and AMD servers under open-source Linux and the freedom to competitively purchase hard and softphones, gateways, switches, etc. from many different manufacturers and vendors. OPEX is reduced by the ability to auto-provision phones, centrally administer the system, ease of use and least cost routing.
To get started exploring sipXcom, you can set up an almost free demo sipXcom system. A public cloud installation for demo and testing can be set up for about $20 per month. A domain can be acquired for as little as $1 for one year. SIP trunking can be set up for $1 per month for a DID telephone number and 1c per minute for incoming and outgoing calls in the USA. Softphones such as LINPHONE are open source and free. Other low-cost options are to install sipXcom on open source VirtualBox or an existing CentOS server.
For firewalls, security and more complex configuration of the system, please check the other wiki sections.
The major steps are:
Create DNS records for your domain
Install CentOS 7
Enter network settings
Enter host and domain settings
Configure server core, telephony, and device services
Configure soft and or hard phones
Add SIP Trunking for incoming and outgoing telephone calls
sipXcom requires correct DNS settings to work. And it requires that your PC's (if you want to run softphones) and SIP devices have DNS working properly. 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.
The following are the required records for a single server sipXcom system.
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 the guidance in the link below:
Download a CentOS 7 ISO from here, burn the image on a physical DVD/CDRom/USB Thumb Drive.
Here's a video to help...
Video of initial configuration...
Log in 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.
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).
Note: Some browsers are disallowing self-signed certificates (IE and Chrome). Try using FireFox and adding in the self-signed certificate as trusted to get by. Then get a certificate added as soon as possible. See: Certificates
You may get a certificate warning as seen below (browser dependent... for Chrome click and then click 'Proceed to XXXXXX (unsafe)'):
Use superadmin for User ID and password set in the step before for logging in Web UI.
Insert one or more external DNS servers that can resolve external names (System Menu -> DNS).
Logout of the Admin Portal. Then sign in to sipXcom server as root.
At root, enter:
after shutdown completes, enter
after yum update completes, always reboot the server:
Guidance on upgrading can be found at:
Log into Admin Portal. Select all the services checked and Apply. You can mouse over the services for a description of each service.
DHCP service supplies devices such as phones with IP addresses and other critical network settings. Only one DHCP service is allowed per cluster and is an optional service. If DHCP is external to the system, then the external DHCP service must have Option 66 enabled to the address of the sipXcom server.
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).
Select all the services checked and Apply. You can mouse over the services for a description of each service.
Select all and Apply.
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.
Softphones such as LINPHONE are usually configured on the client; however, BRIA 3.x softphone can be configured on the sipXcom server.
Hard phones such as Polycom are usually managed by the sipXcom server which sends profiles to the hard phones including boot rom, sip image and line configuration. Many hard phones may be initialed configured with sipXcom auto provisioning or may be manually provisioned.
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 softphone, such as Jitsi or LINPHONE for Windows, MAC OSX and Linux can be used.
LINPHONE can be downloaded from: http://www.linphone.org/
LINPHONE is also available at no cost for:
Mobile Devices – Android, iOS and Windows 8 from Appstore and Google Play
Web Browsers - Chrome, Internet Explorer, Mozilla Firefox and Safari @ http://web.linphone.org
LINPHONE for Windows is configured as follows:
Go to Help and select Account Assistant
Select "I have already a SIP account and just want to use it" and click Forward
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.
At the bottom of the screen, please see "Registration on sip:sip.nycsip.com successful."
Enter SIP Address or PSTN phone number and click LINPHONE symbol at right. In order to dial out to the PSTN, you need SIP Trunking set up. SIP Trunking is covered later.
Dialer reflects call in progress.
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
To automatically configure phones, Server Configuration – Device Provisioning must be set up. Just plug in a supported phone to the VOIP network or VLAN that sipXcom resides on and sipXcom will automatically configure the phone.
Polycom phones will self configure when plugged into the sipXcom system network. The Cisco Discovery Protocol is used. The phone will initially configure with an unique extension ID displayed on the phone such as W9B. The new phone will be listed with the unique ID in the description under Devices, Phones. At this point, you can add lines to this phone, save and send profiles. The phone will re-boot and display the lines added. Please also see http://wiki.sipxcom.org/display/sipXcom/Configuring+Polycom+Soundpoint+IP+Phones
The alternative to the automatic configuration is manual configuration described next.
You can either go into a user and click Phones and then use the 'Add new phone...' dropdown box, or go into Devices -> Phones and find the same 'Add new phone...' dropdown box.
Users -> Select a User -> Phones (left side menu)
Devices -> Phones
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.
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.
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.
Contact an Interoperable ITSP provider to get account information and ITSP server information from list at:
Insert Name of ITSP, use built-in SIP Trunk SBC, use provider template and select from the drop down, and FDQN of ITSP server. Click Apply and OK.
Set up Caller ID
Enter caller id and name. Click Apply and OK.
Click on dial plan to be used with gateway.
Select Long Distance dial plan and then Apply and OK.
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.
You are now ready to Send and Receive Calls from the Public Switch Telephone Network (PSTN).