Uniden UIP1868 SIP Phone

Features

• Includes both wired handset on base unit and 5.8GHz wireless handset
  • Both have speakerphone
• SIP Phone with support for two lines
  • Codecs: G711u, G.711a, G.729, G.726
  • Both lines must use the same server
  • Supports two external analog phones with RJ-11 jacks, one on each line
  • Second line only available on external analog phone or fax
  • Second line may not work! It disappeared from the SIP Settings page after the first day. -Jason
• Integrated broadband router with PPPoE, Port Forwarding, Port Triggering, WAN MAC Spoofing, DHCP with Reservations, ToS/QoS/Basic Traffic Shaping
  • One Ethernet WAN jack and one LAN
• Accepts direct SIP calls, but won't traverse NAT

Drawbacks

• Only dials through configured server, won't dial direct to other SIP phones or gateways (sipbroker)
• No STUN support
• No DDNS support

Phones Locked to One VSP

This device has been sold by multiple VSPs in a locked state. Those versions are called UIP1868P or UIP1868V. The unlocked versions are called UIP1868 and UIP1868G. The information below applies to the generic, unlocked versions, but may apply to all versions. It may be possible to unlock the phone by updating the firmware. Search for DTA-200 firmware.

User Manuals

• User Manual for hardware phone features: http://www.uniden.com/pdf/UIP1868Pom.pdf
• A user manual for VOIP and router features is available. Details are provided inside the package.

ToS/QoS/Traffic Shaping

Current residential internet services have low upload bandwidth, so conflict can easily occur between VoIP calls and other network traffic. The traffic shaping features can eliminate the conflict when properly configured.

1. Determine your upload bandwidth. One way is to run a Speed Test at http://www.dslreports.com/stest.
2. Set the Uplink Bit Rate on the SIP QoS Settings page. For example, on a Comcast cable modem with 350kbps upload, set the Uplink Bit Rate to 350.
3. On the RESET page, choose Reset and execute Main Application.

Miscellaneous

Incoming PPTP VPN works with port 1723 forwarded to an internal VPN server.

If you plan to dial this device directly, make sure it isn't behind a firewall using NAT (like a broadband router). The lack of STUN support means it can't deal with NAT.

To use a SIP server on the LAN side of the UIP1868:

1. Tell your server to listen on a port other than 5060-5061 and 10000-10015. In Asterisk this is bindport in sip.conf
2. Forward the server port through the UIP1868 on the LAN Port Forwarding page
3. Set the UIP1868 SIP Outbound Proxy and Registrar the WAN IP of the UIP and the port forwarded above

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UIP1868 and Asterisk

1. Set SIP Server and Outbound Proxy and Registrar to Asterisk server
2. Set Phone Number to Asterisk SIP devicename (can be numbers, letters, and symbols)
3. Set Password
4. On the RESET page, choose Reset and execute Main Application. Do this anytime you change SIP settings.
5. After the reset, look for Line 1 Status: Register Success on the SIP Status page. The call agent light should be steady. A short ring indicates successful registration, and Incoming Call is displayed on the base unit.

UIP1868 and VoiceMeUp.com

1. Set SIP Server and Outbound Proxy and Registrar to the proxy server (provided in the control panel)
2. Set Phone Number to your chosen username (provided in the control panel)
3. Set Password to the SIP password (provided in the control panel)
4. On the RESET page, choose Reset and execute Main Application. Do this anytime you change SIP settings.
5. After the reset, look for Line 1 Status: Register Success on the SIP Status page. The call agent light should be steady. A short ring indicates successful registration, and Incoming Call is displayed on the base unit.