Dynamix DW 4 FXO Manual do Utilizador descarregar pdf (Página 21)

Dynamix IP PBX-100

you should set the Trunk ID (Trunk Number) in the Trunk page first; then you could input the

correct Trunk ID in the Destination field.

Password: Assign the register password for device to register on IP PBX-100.

Host: Setting the Host to Dynamic will require the trunk to register the IP PBX-100 so that

the IP PBX-100 knows how to reach the trunk. You can also set the Host to an IP address or

FQDN if you set the Host to Address. There will be a field called [Address] appear when you

choose Host to Address. This limits only where you pace case to, as the user is allowed to

place case from anywhere.

DialPlan: Define the dialing pan for Trunk. It specifies the location of the instruction used to

control what the phone is allowed to do, and what to do with incoming case for this

extension. In phase 1, there is only one DiaPan called [greeting].

DTMF: User can select DTMF type to be RFC2833, In-band, or SIP-lnfo.

NAT Traversal: If the Trunk device is behind a device performing NAT, such as firewall or

router, and need to register to IP PBX-100 on public network, then user has to enable this

function. Enable NAT Traversal to force IP PBX-100 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

oftheIP PBX-100.

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 IP PBX-100 if the IP PBX-100 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. (The default SIP signal port is 5060)

External Server Address: This field will a l l o w 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 c a s e could go via this Trunk. Default is no limited.

Outbound Caller ID: Some service provider will require the correct registered caller ID if it

got an incoming cal. Default the IP PBX-100 will send the Extension's caller ID to this Trunk,

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