User guide

Chapter 5 — Phonebook Configuration
Multi-Tech Systems, Inc. 65
Field Name
Values
Description
Total Digits
as needed
Number of digits the phone user must dial to reach specified destination. This field not
used in North America
Remove Prefix
dialed digits
Portion of dialed number to be removed before completing call to destination.
Add Prefix
dialed digits
Digits to be added before completing call to destination.
IP Address
n.n.n.n
The IP address to which the call will be directed if it begins with the destination pattern
given.
Description
alpha-numeric
Describes the facility or geographical location at which the call will be completed.
Protocol Type
SIP or H.323
or SPP
Indicates protocol to be used in outbound transmission. Single Port Protocol (SPP) is a
non-standard protocol designed by Multi-Tech.
H.323 fields
Use Gatekeeper
Y/N
Indicates whether or not gatekeeper is used.
Gateway H.323
ID
alpha-numeric
The H.323 ID assigned to the destination MultiVOIP. Only valid if “Use Gatekeeper” is
enabled for this entry.
Gateway Prefix
numeric
This number becomes registered with the GateKeeper. Call requests sent to the
gatekeeper and preceded by this prefix will be routed to the VOIP gateway.
H.323 Port
Number
1720
This parameter pertains to Q.931, which is the H.323 call signaling protocol for setup
and termination of calls (aka ITU-T Recommendation I.451). H.323 employs only one
“well-known” port (1720) for Q.931 signaling. If Q.931 message-oriented signaling
protocol is used, 1720 must be chosen as the H.323 Port Number.
SIP Fields
Use Proxy
Y/N
Select if proxy server is used.
Transport
Protocol
TCP or
UDP
VOIP administrator must choose between UDP and TCP transmission protocols. UDP is a
high-speed, low-overhead connectionless protocol where data is transmitted without
acknowledgment, guaranteed delivery, or guaranteed packet sequence integrity. TCP is
slower connection-oriented protocol with greater overhead, but having acknowledgment
and guarantees delivery and packet sequence integrity.
SIP Port Number
5060 or other
*See RFC 3087
(“Control of
Service Context
using SIP Request-
URI,” by the
Network Working
Group).
The SIP Port Number is a UDP logical port number. The VOIP will “listen” for SIP
messages at this logical port. If SIP is used, 5060 is the default, standard or “well known”
port number to be used. If 5060 is not used, then the port number used is that specified
in the SIP Request URI (Universal Resource Identifier).
SIP URL
sip.userphone@ho
stserver, where
“userphone” is the
telephone number
and “hostserver” is
the domain name
or an address on
the network
Looking similar to an email address, a SIP URL identifies a user's address.
In SIP communications, each caller or recipient is identified by a SIP URL:
sip:user_name@host_name. The format of a sip URL is very similar to an email address,
except that the “sip:“ prefix is used.
SPP Fields
Use Registrar
Y/N
Select this checkbox to use registrar when VOIP system is operating in the
“Registrar/Client” SPP mode. In this mode, one VOIP (the registrar, as set in Phonebook
Configuration screen) has a static IP address and all other VOIPs (clients) point to the
registrar’s IP address as functionally their own. However, if your VOIP system overall is
operating in “Registrar/Client” mode but you want to make an exception and use Direct
mode for the destination pattern of this particular Add/Edit Phonebook entry, leave this
checkbox unselected. Also do not select this if your overall VOIP system is operating in
the Direct SPP mode – in this mode all VOIPs are peers with unique static IP addresses.