TD 92930EN
13 December 2013 /Ver B
Configuration Manual
WL3 and WL3 Plus WLAN Handset
57
6. Handset Configuration
• Secondary SIP proxy – defines the optional secondary SIP proxy, which is used if
the handset fails to register with the primary SIP proxy. See definition examples in
Primary SIP proxy above.
When the handset has connected to the Secondary SIP proxy, it continuously tries
to reconnect to the Primary SIP proxy.
• Outbound proxy – defines the primary outbound proxy by a domain name, an IP
address, or an IP address together with a port number. The parameter is only
visible, if the parameter Outbound Proxy mode is set to “Yes”.
• Listening port – the port that the handset listens to for incoming SIP traffic.
• SIP proxy ID – defines the SIP proxy by a domain name.
NOTE: This parameter is only needed when an outbound proxy is defined. It can
also be used to specify a domain name when parameters Primary SIP proxy and
Secondary SIP proxy have been assigned IP-addresses.
• SIP proxy password
• Send DTMF using RFC 2833 or SIP INFO – this parameter defines the path the
DTMF signaling should take. If set to “RFC 2833“, the DTMF signaling is sent in the
RTP stream, that is, from handset to handset. If set to “SIP INFO”, the DTMF
signaling is sent using SIP signaling, that is, through the PBX.
• Hold type – defines type of hold to send when the handset puts a call on hold. The
selection depends on what types of hold the PBX support. For more information
about what types of hold the PBX support, see the applicable documentation for the
PBX.
• Registration identity – defines if the endpoint uses its number or ID for the
registration with the SIP proxy.
• Authentication identity – defines if the endpoint uses its number or ID for the
authentication with the SIP proxy.
• MOH locally – Music on hold is played by the handset i.e. if the PBX does not
supply MOH, the handset plays a tone when the call is on hold.
• Hold on transfer – puts a second call on hold before transfer, which is required by
some SIP proxy servers.
• Direct signaling – defines whether calls originating from other sources than the
configured SIP Proxy should be accepted or redirected using “USE PROXY“
message.
• SIP Register Expiration - defines the number of seconds for register expiration to
the PBX.
6.10.3 Codec
A codec encodes a stream or signal for transmission. Codecs are often used in
streaming media applications. This setting defines how to packetize and compress the
sound in a voice call.
1 Select VoIP > General.
2 In the Codec configuration drop-down list, select the applicable codec. The
following are possible:
• G.711 A-law (EU)
• G.711 u-law (US)
•G.729
•G.729A
•G.722
3 In the Codec packetization time configuration drop-down list, select
packetization time to use for speech (value between 10 and 60 ms). Default
value is 20 ms.
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