ZyXEL Communications ISG50 User Manual
Chapter 30 Outbound Trunk Group
ISG50 User’s Guide
484
Each field is described in the following table.
Table 166
SIP Trunk Add/Edit
LABEL
DESCRIPTION
Trunk Name
Type the name of this SIP trunk. This field can be 1-30 alphanumeric characters (A-Z, a-z,
0-9) and underscores (_). The first character must be a letter.
0-9) and underscores (_). The first character must be a letter.
Description
Type the description for this SIP interface. This field can be 0-63 alphanumeric characters
(A-Z, a-z, 0-9), underscores (_), hyphens (-) and spaces. 0 means this field can be left
blank.
(A-Z, a-z, 0-9), underscores (_), hyphens (-) and spaces. 0 means this field can be left
blank.
Representative
Number
Number
Enter the phone number that the called party sees if outgoing calls through this outbound
line group don’t match configured rules. associated with the SIP account for this SIP
trunk. In the full SIP URI, this is the part before the @symbol. You can use 3-40 printable
ASCII characters.
line group don’t match configured rules. associated with the SIP account for this SIP
trunk. In the full SIP URI, this is the part before the @symbol. You can use 3-40 printable
ASCII characters.
SIP Proxy Server
Address
Address
Enter the IP address or domain name of the SIP server provided by your VoIP service
provider. You can use up to 64 printable ASCII characters. It does not matter whether the
SIP server is a proxy, redirect or register server.
provider. You can use up to 64 printable ASCII characters. It does not matter whether the
SIP server is a proxy, redirect or register server.
SIP Proxy Server
Port
Port
Enter the SIP server’s listening port number, if your VoIP service provider gave you one.
Otherwise, keep the default value.
Otherwise, keep the default value.
SIP Register
Server Address
Server Address
Enter the IP address or domain name of the SIP register server, if your VoIP service
provider gave you one. Otherwise, enter the same address you entered in the SIP Proxy
Server Address field. You can use up to 64 printable ASCII characters.
provider gave you one. Otherwise, enter the same address you entered in the SIP Proxy
Server Address field. You can use up to 64 printable ASCII characters.
SIP Register
Server Port
Server Port
Enter the SIP register server’s listening port number, if your VoIP service provider gave
you one. Otherwise, enter the same port number you entered in the SIP Proxy Server
Port field.
you one. Otherwise, enter the same port number you entered in the SIP Proxy Server
Port field.
Service Domain
Select Define Service Domain and enter the SIP service domain name. In the full SIP
URI, this is the part after the @ symbol. You can use up to 64 printable ASCII Extended
set characters. If you choose not to use a service domain, select Disable.
URI, this is the part after the @ symbol. You can use up to 64 printable ASCII Extended
set characters. If you choose not to use a service domain, select Disable.
Outbound Proxy
Select Define Outbound Proxy and enter the IP address or domain name of the
outbound proxy server provided by your VoIP service provider. You can use up to 64
printable ASCII characters. It does not matter whether the SIP server is a proxy, redirect
or register server. If you choose not to use an outbound proxy server, select Disable.
outbound proxy server provided by your VoIP service provider. You can use up to 64
printable ASCII characters. It does not matter whether the SIP server is a proxy, redirect
or register server. If you choose not to use an outbound proxy server, select Disable.
Outbound Proxy
Port
Port
Enter the outbound proxy’s listening port number, if your VoIP service provider gave you
one. Otherwise, leave it as the default ‘5060’.
one. Otherwise, leave it as the default ‘5060’.
If the outbound proxy is disabled, then this port will be ignored.
DTMF Mode
Control how the ISG50 handles the tones that the IP phones on your network make when
they push their buttons. One use of the tones is to distinguish between numbers when
trying to dial a PSTN phone number. You should use the same mode as your VoIP service
provider. The choices are:
they push their buttons. One use of the tones is to distinguish between numbers when
trying to dial a PSTN phone number. You should use the same mode as your VoIP service
provider. The choices are:
•
info - Send the DTMF tones in SIP messages.
•
rfc2833 - Follow the RFC 2833 standard and send the DTMF tones in RTP packets.
•
inband - Send the DTMF tones in the voice data stream. This works best when you
are using a codec that does not use compression (like G.711). Codecs that use
compression (like G.729) can distort the tones.
Enable Privacy
Select this to replace the caller’s name and number with “Anonymous”. For example,
“Anonymous”<Anonymous@172.1.1.253>.
“Anonymous”<Anonymous@172.1.1.253>.
Clear this to display caller ID as displayed in this screen’s CallerID Viewer field.
Proxy Require
Enter this (0~127 alphanumeric characters) to inform the SIP server that this device is
behind a firewall or NAT device. Fill this field in only if you were given information by your
SIP service provider.
behind a firewall or NAT device. Fill this field in only if you were given information by your
SIP service provider.
Channel-limit
Specify the maximum number of SIP calls (1~128) allowed to be made through this trunk
connection at one time.
connection at one time.
Session Timer
Enable Session
Timer
Timer
Select this activate the ISG50’s session timer. If you encounter connectivity issues with
your network or Internet, then it is suggested that you use the session timer.
your network or Internet, then it is suggested that you use the session timer.