AMX SIP Communications Gateway CSG-500 - Manual (Page 40)

Telephone System Configuration
32 CSG SIP Communications Gateway Operation/Reference Guide
Create New User Page Options (Cont.)
Line Number Polycom brand VoIP phones are capable of servicing 1 to 6
separate VoIP phone lines, depending on the model of the phone.
If you are using the Polycom Auto-provisioning feature of the
CSG, this option can be used to define which line of your phone
will be used by the user. No more than one user can be assigned
to a line on a phone.
Note: Each phone must be configured with a user that has Line
Number set to 1. Additionally, assigned line numbers must be in a
contiguous range.
Line Keys Polycom brand VoIP phones include multiple line keys. The
number of line keys available will depend on the model of the
phone. If you are using the Polycom Auto-provisioning feature of
the CSG, this option can be used to define how many line keys on
the phone should be associated with this user (e.g. Let’s says you
configure a single Polycom phone with two users. User 6000 with
Line Number set to 1 and Line Key set to 2 will display user 6000
on line keys 1 and 2 on the phone. User 6001 with the same
MAC, Line Number set to 2, and Line Key set to 4 will display
user 6001 on line keys 3, 4, 5 and 6 on the phone.). Be sure not
to select more line keys than your phone supports.
SIP/IAX
Password The password used if the user has a SIP/IAX account.
NAT Try this setting when your CSG is on a public IP, communicating
with devices behind a NAT device (broadband router). If you have
one-way audio problems, you usually have problems with your
NAT configuration or your firewall's configuration of SIP and RTP
ports.
Can Reinvite By default, the CSG will route the media streams from SIP
endpoints through itself. Enabling this option causes the CSG to
attempt to negotiate the endpoints to route the media stream
directly. It is not always possible for the CSG to negotiate
endpoint-to-endpoint media routing. This option can be used to
tell the CSG whether or not to issue a reinvite to the client.
DTMF Mode Set the default DTMF mode for sending DTMF (touch tone). The
default setting is rfc2833. Other options include:
• info - Used to display SIP Info messages
• inband - Inband audio (requires 64 kbit codec - alaw, ulaw)
• auto - Use rfc2833 if offered, inband otherwise.
Insecure Insecure is a SIP parameter used to determine peer matching.
The setting determines whether or not an insecure connection will
be allowed, or if authentication is required. The valid options are:
• port - Enter this value to match against only an IP address. This
setting is useful if you have multiple endpoints behind a NAT
device.
• invite - Enter this value to match against both the IP address
and port number provided in the Contact field of the SIP
header. A call will be allowed without authentication if a match
is found.
• very - Specify this value if you do not want to require
authentication upon an initial invite.
• no - Specify this value if you do not want to allow an insecure
connection.
3-Way Calling Allows the extension to receive a call and then dial out to another
phone number to conference with the inbound call and the
recipient of the outbound call.
In Directory Check this option if you want a user to be searchable using the
system telephone directory.