Teletronics IP-PBX Server - Manual (Page 15)

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 DTMF: User can select DTMF type to be RFC2833, In-band, or SIP-Info.
 NAT Traversal: If the Trunk device is behind a device performing NAT, such as
firewall or router, and need to register to EZLoop® IP-PBX Enterprise SIP Server on
public network, then user has to enable this function. Enable NAT Traversal to force
EZLoop® IP-PBX Enterprise SIP Server to ignore the contact information for the
Trunk and use the address from which the packets are being received.
 RTP Mode: User can choose for two type of RTP mode, one is Routed Mode another
is Direct Mode. The voice media will be routed “Peer-to-Peer” if two clients are both
setting to Direct Mode. This way will improve the voice quality and reduce the
performance wastage of the EZLoop® IP-PBX Enterprise SIP Server.
Note:
•
If one peer set to Direct Mode but another peer set to Routed Mode, the result
will become to Routed Mode.
•
Voice media will still go through the EZLoop® IP-PBX Enterprise SIP Server if
the EZLoop® IP-PBX Enterprise SIP Server needs to detect DTMF.
•
If you enable the NAT Traversal function for Extension, the RTP mode will
change to Routed Mode directly; this way will avoid the “one-way voice” or “no
voice issue” of VoIP.
•
If the both peers are under different subnet, or one peer is under Public IP but
another one is under Private IP, we strongly suggest you to set the RTP
mode to Routed Mode to avoid some unexpected voice problems.
 Port: You can use this to define the SIP signal port if you want to listen on a
nonstandard SIP signal port.
 External Server Address: This field will allow you to set the domain in the SIP From
URI. Setting this will avoid some unexpected issue if the service provider needs this
for authentication.
 Maximum Channels: This will limit the maximum channels for this Trunk. For
example, you set 2 into this field; only 2 outgoing calls could go via this Trunk. Default
is no limit.
 Outbound Caller ID: Some service provider will require the correct registered caller
ID if it got an incoming call. Default the EZLoop® IP-PBX Enterprise SIP Server will
send the Extension’s caller ID to this Trunk, if you set empty here.
Note:
•
Normally, SIP From URI will contain the Extension’s calling ID and EZLoop® IP-
PBX Enterprise SIP Server’s IP address, but some ITSP may reject this call due
to some security issue. You can modify the Calling ID and IP/ Domain in the
fields of [External Server Address] and [Outbound Caller ID] when the call is
going via the EZLoop® IP-PBX Enterprise SIP Server to the Destination (Trunk)
to avoid such security issue.
• If you set a FXO gateway as the Trunk, you can just use the default Trunk 888
and 889 as the FXO’s register number.
•
For the FXO gateway, you may just only configure Trunk Number, Password,
Host, DialPlan, Keypad, NAT Traversal and RTP Mode.
•
If you set the ITSP as the Trunk, you may need to set the following configure:
Port, External Server Address and Outbound Caller ID.
•
For more information, please refer to the CH5. Appendix-Application
between CPE device and EZLoop® IP-PBX Enterprise SIP Server.
 Hot-Key Tran: Enable this feature will permits the calling party or called party to
transfer a call by pressing the *0 (For Blind Transfer) or *9 (For consultant