27
Parameter
(Single Server Mode)
Description
Default Value
1. Authentication ID
The name of the device used in Caller Identification is defined here.
2. Password
The password used for SIP registration is specified here.
3. SIP Proxy
The domain name or IP of the SIP Proxy or Server is specified here. If the SIP Proxy is
using the standard 5060 signaling port, then there is no need to specify the port
number. Otherwise, the port number can be specified by adding ":" and then the port
number at the end of the SIP Proxy address.
4. SIP Registrar
The address of the SIP Registrar Server is specified here.
5. Re-register Period (s)
Register to the SIP Server at an interval specified by this parameter.
60
6. Phone Number
Normally Phone Number is the same as the Authentication ID. If this parameter is not
set, it is automatically assigned as the Authentication ID by default. Only enter this
field if the Phone Number assigned is different from the Authentication ID.
7. Display Name
This parameter gives a name reference to the Phone Number. If it is set, the Display
Name is to the SIP Server in the SIP INVITE message.
8. Outbound Proxy
The address of the Outbound Proxy used for VoIP communication is specified here.
9. Home Domain
Home Domain is used in SIP identification. It should be specified as required.
10. Backup Server
SIP Proxy
SIP Registrar
11.
Home Domain
Backup Server improve service reliability and is used only when the primary server fails.
This specifies the backup SIP Server address.
This specifies the backup SIP Registrar Server address.
This specifies the backup Home Domain address.
Disabled
12.Routing Prefix
This parameter is used for call routing. When this is set, the corresponding channel is
only used to dial out a phone number with the matching Routing Prefix.
Syntax:
<Prefix1>,<Prefix2>,<Prefix3>,.....
where Prefix is a text string which consists of digits, alphabets, and special characters.
The maximum length of the Routing Prefix is 120 characters.
2. Config. By Line (for all models except GoIP-1) Mode
This mode is only applicable for multi-line models.
Each line (associated with a corresponding GSM
channel) registers to a SIP server separately and operates as an independent phone line. A Routing Prefix
for each channel must be assigned in order to enable the channel to allow making outgoing calls. This
allows to the SIP server to assign which channel to dial out the call for termination. If a channel does not
have its Routing Prefix set, this channel will not be used to dial out any calls. In this mode, the prefix of
the phone number to be dialed out must match one of the Routing Prefixes assigned. The channel with
the matching Routing Prefix will be used to dial out the call. If no match is found, the call will not be
dialed out and a SIP 404 message is returned to the SIP Server. If a match is found but no channel is
available to dial out the call, a SIP 503 message is returned to the SIP Server. The syntax for the Routing
Prefix is defined in the Parameter Table for Single Server Mode.
It is important to note that the Routing Prefix (P) must be removed via the dial plan before the number is
dialed out. The dial plan syntax to remove the Routing Prefix is "P:-P|". Please refer to section 3.3.6 for
more information on the Call Out Dial Plan.
Summary of Contents for GoIP
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