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-User ID: The SIP caller ID will be used according to the
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 header's 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 header's 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
FXO_Tel_NO@local_IP_address.
The following guideline could be used for most of case:
1. If the 25xx in SIP proxy was handled as a gateway,
please set both the display name and User Id to be
"PSTN caller ID".
2. If the Dynamix 25xx in SIP proxy was handled as
a subscriber, please set the display name to "PSTN caller
ID" and user ID to "SIP User ID".
For DNIS is Register TEL: Using FXO port that have an incoming call
from VOIP dial to PSTN, it can be chosen 2 methods. ( FXO only use )
1 stage dialing: If match a sip trunk number or not, it will
direct to dial out that number before Making DM and route
plan.
2 stage dialing: If match a sip trunk number, it will auto
answer the call and playing dial tone.
Keep Alive: Enable or Disable it.
Keep Alive Time (sec): Specify interval time to send sip register
message to proxy server.
FROM
header's
52
URL
will
be
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