Unpack the phone components and cables and plug them together as follows (easiest done with the phone base upside down as shown).
- Click the stand to the base unit.
- Connect the handset using the curly phone cable.
- Connect the SW socket to a free port on your router or switch, using the supplied ethernet cable, this phone is PoE Enabled
- Optional - If required connect a Cisco approved power adapter or if you are short on switch ports then connect your PC to the PC port.
Switch the power on and leave the phone to initialise for two minutes.
- If you have your network set up to hand-out IP addresses (using DHCP), the phone will pick up an IP address and be ready for use. If not, see Setting an IP address
- If you bought your phone from Gradwell, it will already be configured for use with your first extension. If you bought it elsewhere, see Configuring the phone
To answer a call: pick up the handset, or use the Headset button on the headset (if any), or press the Speaker button for hands-free operation.
To switch between two calls on different lines: press the corresponding Line button to put the current caller on hold, and press the other Line button to speak to the other caller. The ‘active line’ is shown by a steady red light and ‘held’ by a flashing light. Unused lines are green.
To hold a call without speaking to another caller: press the Hold (hand) button.
- To dial an IP address, hold the hash key down to enter a dot
- To dial a SIP URL, press 1 then use the down-arrow on the menu navigation key to toggle between lower-case, capital letters and numbers, as indicated in the display. You’ll find the left-arrow key (to backspace and correct mistakes) useful until you get the hang of this! Use the 1 key twice to get the @ character
To end headset/hands-free calls: press the Headset or Speaker button again.
To make attended transfers: put the caller on hold by pressing the xfer softkey. Dial the destination number as usual and speak to the recipient. Then press the xfer soft key again to put the caller through and hang up your connection.
To make conference (three-way) calls: If only one party is currently connected, press the conf softkey to put it on hold. Dial the third party, and when they answer, press the conf softkey again to join all three parties into a conference call.
If both parties are already connected/on hold, press the confLx soft key. If one call is on hold, the conference call starts immediately. If more than one call is on hold, the phone prompts you to select the other line key to join with the active call.
To pick up voicemails: dial *1 and enter the voicemail password when prompted (which you will find in the extension details in your VoIP control panel).
To make calls: enter the number using the keypad then pick up the handset to make a call. Press the dial softkey at the left bottom of the display to start dialing. Press the Headset or Speaker button before dialing for a headset or hands-free call, respectively.
Advanced configuration and troubleshooting
Setting an IP Address
From the phone keypad:
- Press the Setup button (folded page icon)
- Select option 9 for network settings
- Select option 1, Select y/n to set to off, press the OK key.
- Select option 8 to set the IP address (use the * to enter the full stop)
- Select option 9 to set the subnet mask.
- Select option 10 to set the default gateway.
Configuring the phone
(Only necessary for phones not purchased from Gradwell).
- Press the Setup Key (folded page Icon Key)
- Select option 9 for network settings
- Make a note of the IP address
Type the IP address into your browser like this: http://nnn.nnn.nnn.nnn/admin/advanced (replace each nnn by as many digits as appear in the phone display, for example http://192.168.0.120/admin/advanced), then press Enter, you should now see the Cisco webpage showing the status of the phone.
Click the Ext 1 tab at the top.
The important fields here are all provided in your welcome email or the control panel for the first (default) extension:
- Proxy and Registration section: Proxy (the SIP proxy server) and also the Outbound NAT proxy server (if required)
- Subscriber info section: User ID and Auth ID (your first Centrex extension number), and Password (your account password set in the Extensions control panel).
Click Submit all Changes and the phone will restart and register with the service provider.
By Default we support the G711a or G711u Codecs but if you are using multiple phones on your broadband connection, or you have a low bandwidth internet connection, then in the Audio Configuration section we recommend you choose G729a as the Preferred Codec. This will allow the phone to use less bandwidth where possible.
Please note that if you wish to use G729a you will need to request that this codec is enabled for your extension.
Please set your DTMF Tx Method to AUTO this will allow our system to recognise what keys you press on your phone.
If you intend to use multiple Cisco SPA IP telephones behind a Nat Firewall you will need to give each port on the Cisco SPA telephones a unique SIP Port number.
For example if you have two Telephones and you wish to use all four lines then on the first Telephone you would set ext1 sip port to 5060 and ext2 sip port to 5061, on the second Telephone you would set ext1 sip port to 5062 and ext2 sip port to 5063.
One way audio between phones on the same network
We have had reports of Cisco phones that successfully make and receive calls that originate from outside the network, but give one way audio issues on internal calls to other makes of handset.
If you have ensured that you have correctly configured your router and still have this problem, one of our customers resolved the issue by undertaking the following steps:
- Changing the RTP range for the phone to only be 5004 (as opposed to the default range)
- Set the device to have a static internal IP address and a subnet mask of 255.255.255.0
- Removed all codecs aside from G711u