2N NetStar Admin manual - Manual (Page 64)

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SIP
Always mediate RTP – if this option is checked off, the RTP stream is always
routed through the PBX VoIP card. If not, the RTP stream is processed outside
the PBX (in the case of VoIP – VoIP connection) and the PBX is reponsible for
signalling only.
Reverse RTP negotiation – use this option to define the way of codec
negotiation. If this option is not checked, codecs are offered already in the
message. Invite
Use short headers – select this option to enable using short headers (e.g. From
= f, To = t, Via = v) for data transfer minimisation.
Do not replace +,#,* in numbers – these characters are replaced with
corresponding strings %xx in numbers only if the option is not selected. If it is
selected, they are sent.
Route by To header – if this option is selected, the incoming calls of the given
port are routed by the
header. In other cases (including the default setting!), To
the calls are routed according to the header. Request URI
Dedicated Registrar – use this option to enable routing registrations from
gateways to another destination.
Address – the selected Registrar server address.
Port – the selected Registrar server port.
Scheme – set the sip or tel scheme in the "To" and "From" headers from the
SIP. The tel selection is used for networks based on the numbering plan
according to the E.164 recommendation.
Min. MTU – set the packet size limit for obligatory TCP use in the UDP&TCP
mode. The recommended maximum value is 1,448 bytes.
Headers
Complete domains – use this section to specify the domain to be used for the
From and To headers.
Send information
– P–Asserted–Identity – select this item to activate the
P–Asserted–Identity header in the INVITE message for CLIR transmission. The
opponent can thus obtain CLI information even if its CLIR (Calling Line
Identification Restriction) is enabled. By default, CLIR is enabled on the SIP
Gateway port (header active) and disabled on the SIP Proxy port (header
inactive).
Aliases
This section enables to define further Realms(Domains) to be accepted. Those
incoming calls (their INVITEs) will be accepted to the port whose Request–URI field
shall match the SIP gateway or SIP proxy Domain and Alias settings.