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OVERVIEW
This quick start guide covers basic steps from installing sipXecs to placing internal and external calls. Whether you are installing thousands of phones or just setting up a demo system, sipXecs graphical user interface makes the process straightforward and easy.
sipXecs scales to tens of thousands of users and has high availability features required for large installations. sipXecs has extensive features necessary to replace legacy telephone systems and includes significant unified communications capabilities.
For large and complex installations, sipXecs 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 sipXecs running on standard Intel and and AMD servers under open source Linux and the freedom to competitively purchase hard and soft phones, 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 sipXecs, you can set up an almost free demo sipXecs 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 USA. Softphones such as LINPHONE are open source and free. Other low cost options are to install sipXecs 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
Create ISO DVD and boot from ISO DVD
Enter network settings
Enter host and domain settings
Update sipXecs with maintenance releases
Select server core, telephony and device services
Add Users
Configure soft and or hard phones
Add SIP Trunking for incoming and outgoing telephone calls
DNS Records for Your Domain
sipXecs 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.
The following are the required records for a single server sipXecs 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 guidance in link below:
http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs
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 sipXecs and all related compenents.
WARNING: The installation will delete ALL data on ALL drives connected to the system. REALLY, IT WILL! It will format the hard drives, create its own partitions and the process will start as soon as you hit Enter. Make sure that you know what you are doing..
Insert the DVD into your Intel or AMD server and power it up.
Installation Wizard
- Press enter on the boot screen to begin sipXecs installation
On Configure TCP/IP screen, select Manual Configuration under Enable IPv4 support and then select OK.
On Manual TCP/IP Configuration screen, set the static IPv4 address and subnet mask of the sipXecs server, gateway and name server and click OK.
Select the language, keyboard and time zone to be used during the installation.
Set a root password for sipXecs server.
Click Next on Boot Loader screen.
After CentOS finishes installing, check Next.
Network Settings
Login to the system as root with the password you provided earlier and continue to configure sipXecs 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 displayed by the setup wizard (just the hostname of your server).
- 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.
Update sipXecs with maintenance releases
Logout of the Admin Portal. Then sign in to sipXecs server as root.
At root, enter:
service sipxecs shutdown
after shutdown completes, enter
yum update
after yum update completes, enter
service sipxecs start
Guidance on upgrading can be found at:
http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Version
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.
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 sipXecs 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).
| Anchor | ||||
|---|---|---|---|---|
|
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.
Server Profiles
Send profile changes to services.
Add Users
Click "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.
Configure Phones
Softphones such as LINPHONE are usually configured on the client; however, BRIA 3.x softphone can be configured on the sipXecs server. Details of configuring LINPHONE are provided in the soft phone configuration below. Please see following link for Bria configuration:
http://wiki.sipfoundry.org/display/sipXecs/Basic+Bria+Configuration+with+Sipxecs+4.6
Hard phones such as Polycom are usually managed by the sipXecs server which sends profiles to the hard phones including boot rom, sip image and line configuration. Many hard phones may be initialed configured with sipXecs auto provisioning or may be manually provisioned.
Softphone Configuration
An open source SIP soft phone, LINPHONE for Windows, MAC OSX and Linux 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
Click Forward
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 sipXecs and you are ready to make calls between extensions.
Username: must be a User ID already set up on sipXecs
Password: SIP Password for the above Username from sipXecs 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."
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.
Hard Phone Configuration
- Navigate to Devices > Phones
ADD New Phone
sipXecs 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 several places including
http://wiki.sipfoundry.org/display/sipXecs/What+Phones+to+Choose and
http://wiki.sipfoundry.org/display/sipXecs/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 sipXecs resides on and sipXecs will automatically configure the phone.
Polycom phones will self configure when plugged into the sipXecs 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.sipfoundry.org/display/sipXecs/Configuring+Polycom+SoundPoint+IP+Phones
The alternative to automatic configuration is manual configuration described next.
- In the Add new Phone drop down select the phone model you are going to use (Polycom SoundPoint IP 550 in this example)
Insert the serial number of the phone which is the MAC address. Also select the most current firmware version available on sipXecs for this model of phone.
For Polycom phones, please see the firmware matrix for supported firmware at
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html.
For VVX models, please see
http://downloads.polycom.com/voice/voip/uc_sw_releases_matrix.html
These matrices show firmware compatibility for each Polycom phone model.
For Polycom Soundpoint IP 550, the latest firmware supported by Polycom is 4.1.1 but it can run on prior versions.
Check Wiki FAQ for latest guidance. For example, see FAQ for release 4.6 which specifies 4.1.5 split for Polycom VVX 500 and VVX 600 and 3.1.3 for legacy Polycom Soundpoint IP phones.
http://wiki.sipfoundry.org/display/sipXecs/4.6+FAQ
Click OK.
Add New Lines
- After creating the phone navigate to User Lines tab and Add New Line
- Select the phone to be assigned to user 200.
Perform the same steps for the second user - add new phone and assign user 201.
Send Profiles
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.
Place an internal phone call
- After sending profiles the phones will reboot and you can place a call from extension 200 to 201.
SIP TRUNKING
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:
http://wiki.sipfoundry.org/display/sipXecs/Interoperable+Providers
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.
As an example of how to configure SIP trunking, please see http://myitdepartment.net/blog/191
Incoming Calls
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).