User guide

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No set on FXO lines.
-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 FROM header’s URL will be
FXO_Tel_NO@local_IP_address.
The following guideline could be used for most cases:
gateway, please set both the display name and User Id to
be “PSTN caller ID”.
a subscriber, please set the display name to “PSTN caller
ID” and 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. ( FXO
port dialing out only )
 1-stage dialing: When there is an SIP trunk incoming call to
directly without doing DM and routes plan directly.
Note: If
incoming call from VoIP or FXO port Only route to
FXS port. However, the outgoing call from FXS port
go to either VoIP or FXO port depend on DM and
routes plan.
 2-stage dialing: When there is an SIP trunk incoming call to
trunk to wait for SIP trunk user to dial digits and send these
digits to FXO/PSTN network one by one.
 Keep Alive: Enable or Disable it.
 Keep Alive Time (sec): Specify interval time to send SIP register
message to proxy server.
1. If the Dynamix 2522 in SIP proxy was handled as a
2. If the Dynamix 2522 in SIP proxy was handled as
Dynamix 2522, it selects an free FXO port and dial out digits
Dynamix 2522 was configured to PABX Mode, the
Dynamix 2522, it answers this call and play dial tone to SIP