Code Blue IP1500 VOIP SPEAKERPHONE - Manual (Page 24)

Code Blue • 259 Hedcor Street • Holland, MI 49423 USA • 800.205.7186 • www.codeblue.com GU-137-E page 24 of 66
IP1500 and IP2500 Series
Administrator Guide
Configuring a SIP Account
Either of the speakerphone’s two accounts can be configured to register to a VoIP system via SIP.
Configuration is as follows:
• Set the VoIP Protocol to SIP & RTP.
• For Description, enter a name the speakerphone will use internally to refer to this account.
• For Username/Number, enter the number that the speakerphone will use for SIP addressing.
This will often be the extension number in a VoIP-based PBX.
• For Display Name, enter the display name the speakerphone will send in SIP transactions. This
will often be the calling name of the extension.
• For Domain, enter the domain the speakerphone will register to.
• For Outbound Proxy, enter a SIP proxy the speakerphone should send outbound calls to. If this
is the same as the domain, you can leave this field blank.
• For Outbound Proxy Port, enter an IP port number the speakerphone will send outbound calls
to. Typically, this should be left at 0.
• For Registration Lifetime, enter the time in seconds the speakerphone will request that its
registration be valid for. The speakerphone will automatically re-register before this time period
expires.
• Check Keep-Alive if you want the speakerphone to periodically send OPTIONS requests to the
SIP server, e.g. to keep a NAT connection alive.
• Check STUN if you want to enable STUN support for this account.
• You can adjust the DTMF Threshhold value if you have difficulties with the speakerphone
activating in-call commands when no
DTMF is present.
• For Username and Password, set
the username and password the
speakerphone will use to authenticate
to the domain and outbound proxy.
Note that the username is used for
authentication only and need not
match the Username/Number field if
the VoIP system does not expect it to.
• VLAN user priorities can be adjusted
for SIP and RTP audio.