Siemens HiPath optiPoint 410 S Setup Manual page 5

Name of the realm. Within a realm certain combinations of user & password are valid.
During registration the Asterisk server assigns the realm name to the telephone. The pre-
setting is asterisk.
•
bindport
UDP port the asterisk server uses for receiving data. Pressetting: 5060.
•
bindaddr
IP address used to receive incoming SIP connections. Using the pressetting 0.0.0.0 As-
terisk receives SIP connections on all network interfaces and aliases.
•
allow
Permitted codecs. Only the codecs listed here are permitted. The order of the allow en-
tries determines the preference of the codecs.
•
qualify
If the value yes is used, you can monitor latency and availability of the end point. As per
presetting, an end point is considered to be available if the latency is below 2,000 ms. In-
stead of yes/no you can also specify a maximum latency in milliseconds.
•
dtmfmode
Using DTMF (Dual Tone Multiple Frequency) tone signals to control the connection are
sent via the keypad. These are used in menu-driven info services (Attendant Control), e.g.
the voicemail features of Asterisk. There are 3 options to transmit the signals1) inband: the
tone signal is faded into the voice signal; 2) outband: another data stream is started for the-
transmission of the tone signal; 3) instead of a tone signal an INFO field is sent. optiPoint
telephones support both inband as well as INFO fields. We recommend to use INFO fields
as in case of using inband and a high degree of compression there might be errors in the
data interpretation. Set dtmfmode to info.
The following section explains the entries for the individual user agents.
30. June 2006
optiPoint 410/420 - Using with Asterisk, Configuration Guide
Configuration of Asterisk
SIP Registration
c01.fm
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