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.
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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.
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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
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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).