S E S S I O N I N I T I A T I O N P R O T O C O L ( S I P ) C O N F I G U R A T I O N
V I A W E B
Rev H
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If the RFC2833 protocol for Telephony events is enabled, voice packets can be suppressed
during the transmission of the RFC2833 Telephone Event packets.
The SIP RTP Telephone Event Signaling (RFC2833) configuration parameters are configured
per line.
Another alternative is to send the DTMF via the SIP INFO method.
16.3.2 Call Forward Configuration
To set the Call Forward Configuration:
•
Select the setting of the Call Forwarding feature. The Call Forwarding applies during call
establishment by providing a diversion of an incoming call to another destination alias
address. Call forwarding can also be activated by dialing *1 or *2 and deactivted by
dialing *3 (See link 4.4.3). The following stetting are available:
°
Disabled
°
Conditional
Callees frequently wish to redirect incoming calls to an alternative destination if the
primary destination fails to answer within 20 seconds. The reasons for failure are
multifold. They may include busy callee, disconnected callee’s phone, user who
currently does not answer, or user denying the incoming call. The alternative
destination is typically a voicemail system but it may be also another human or some
other SIP device.
°
Unconditional
The call is always diverted to another destination.
Both types of Forwarding, Conditional and Unconditional can be selected.
16.3.3 Gain Control Configuration
To set the Gain Control Configuration (headset volume) for VoIP calls:
•
In the SIP page horizontal menu bar, select
Line1
or
Line2
.
°
Select a
Line (input) Gain
value in the range of <-12 to +3>. This is the volume at
which the callee will hear the caller.
°
Select a
Headset (output) Gain
value in the range of <-12 to +3>. This sets the
volume at which the caller will hear through the headset. If this volume is too loud, an
echo will occur in the headset.
°
Click
Save Line Settings
to effect the changes.
16.4
Phone Book Configuration
•
A direct SIP call may be placed without the need to register on a SIP Server. Use the
Phone Book selection in the SIP application as follows to enable direct calls:
°
In the SIP page horizontal menu bar, select
Phone Book (see
. Notice
that up to 9 phone book settings are available.
°
Enter the URL data - which includes the user information, host and port- for each call
(up to 9). The format is: <userinfo>@<host>:<port>
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