User guide

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SIP display name: The display name will be the Display
Name set in this SIP trunk.
FXO Tel NO: The display name will be the incoming FXOs TEL
No set on FXO lines.
User ID: The SIP caller ID will be used according to th e
following type.
SIP user ID: If the SIP user ID is set, the SIP user ID set in
this SIP trunk will be used and the domain/SIP proxy will be
the host part. The SIP FROM headers URL will be the
SIP_User_ID@Domain or SIP_User_ID@SIP_Proxy_Server.
PSTN caller ID: If the PSTN caller ID will be used in SIP URL,
the SIP FROM headers URL will be
PSTN_Caller_ID@local_IP_address.
FXO Tel NO: If the FXO Tel NO will be used in SIP URL, the
SIP FROM headers URL will be
FXO_Tel_NO@local_IP_address.
The following guideline could be used for most of case:
please set the display name and User Id should be set to
PSTN caller ID.
subscriber, please set to the display name to PSTN caller
IDand user ID to SIP User ID.
For DNIS is Register TEL: When you have a call from VoIP to FXO to
call out to PSTN network, there are two methods can be used.
Keep Alive: Enable or Disable it.
Keep Alive Time (sec): Specify of times send sip register message to
proxy server.
1. If the Dynamix 2540 in SIP proxy is handled as a gateway,
2. If the Dynamix 2540 is SIP proxy is handled as a