AWB OD200-IDU, Руководство оператора

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VoIP Settings
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packet routing. The problem of NAT firewalls can also be solved using a proxy
server to control SIP traffic. Specify the IP address and TCP port used by the STUN
server. (Default: 0.0.0.0:3478, “0.0.0.0” means not available; Port Range: 0-65535)
DTMF – Enables the sending of dual-tone multi-frequency (touch tone) phone
signals over the VoIP connection. There are several methods to choose from:
•
No DTMF: The DTMF signals are not sent over the VoIP connection.
• In-band Mode: The DTMF signals are sent over the RTP voice stream. In the case
when low-bandwidth codecs are used, the DTMF signals may be distorted.
• 2833 Relay: Uses the RFC 2833 method to relay the DTMF signals over the RTP
voice stream without any distortion. (This is the default.)
• Both In-band and 2833: Uses the best method depending on the codecs selected.
Invite Timeout – The time that the unit waits for a response to a SIP Invite message
before a call fails. If network connections are slow and many SIP calls fail, you may
need to increase this timeout value. (Range: 1-300 seconds; Default: 12 seconds)
T.38 Option – Selects the method to use when sending fax messages over the VoIP
network from a fax machine connected to one of the RJ-11 Phone ports on the
OD200. (Default: Voice and T.38 Fax Relay)
•
T.38 Fax Relay: The SIP protocol sets up the VoIP call, then the T.38 Fax Relay
protocol sends the fax data over the network.
• Voice and T.38 Fax Relay: Enables voice calls and faxes to be sent from the
Phone port connection. When a fax tone signal is detected on the port, the T.38
Fax Relay standard is used instead of the voice codec.
• Voice and Fax Pass Through: Enables voice calls and faxes to be sent from the
Phone port connection. For this option, fax signals are sent over the VoIP network
using the voice codec, just as if it were a voice call.
Dial Plan
A dial-plan string can be specified to control phone numbers dialed out through the
OD200. A dial plan describes the number and pattern of digits that a user dials to
reach a particular telephone number. Access codes, area codes, specialized codes,
and combinations of the number of digits dialed can all be part of a dial plan. This
enables a user to predefine dialling sequences that are permitted. It can help
transfer, check, limit phone numbers, and handle prefixes to certain numbers.
The dial-plan string consists of a single digit rule. A typical example of a dial-plan
string is: [0123]xxxxxx.t
Three standard dial plans are defined; Call Transfer Key, New Call Key, and 3-way
Conference. Up to 10 other dial plans can be defined by the user.