UTT-110B VOIP Gateway (SIP)
User manual
Copyright
©
2013, Shenzhen Ultrative Technologies Co., Ltd All rights reserved
21
Asterisk:
When
soft switch as
Asterisk, optional for this option.
VOS type
When the server mode selection VOS
encryption, media
encryption or choose optional signaling encryption.
The primary server
Configure the primary SIP server's IP address or
domain
name.
Primary server port
Configure the primary server SIP register
port, the default
value 5060.
Backup server
switch
Standby server
switch,
the default is off, you need to use an
alternate server is turned on.
Standby server
After the
standby server
opens, enter the backup SIP server's
IP address or domain name.
Standby server port After opening the backup
server,
enter the alternate port SIP
registration server.
Domain
Fill sip server's
domain, under normal circumstances, and fill the
same SIP server IP address for the domain; butt IMS, IMS platform
to fill in the domain name.
Local signaling port
This equipment SIP
signaling port, the default value 5060.
Registration refresh
time
SIP registration refresh
time, in seconds, the default value
600, the actual registration refresh time and softswitch negotiation.
rport
The default value is
off, select Open with rport required field.
Heartbeat switch
Heartbeat
off,
do not send a heartbeat message to the SIP
server; heart open, it will send option heartbeat information to the
SIP server.
Heartbeat interval
Send
heartbeat interval, the default value of 0 seconds, can be
modified.
Heartbeat Timeout
Send
heartbeat timeout, during which time the scope of this SIP
server if the response has not been the heartbeat information that
has been disconnected from the server, the default value of 30
seconds, and can be modified.
PRACK
When turned on, invite support 100rel; closed, invitedoes not
support 100rel
Session Update
When turned
on,
support UPDATE; when closed, does not
support UPDATE
Session Update
After the update session is
open,
the session update the
default value is 360, the actual conversation time updates and
softswitch negotiation.
SIP URI parameter
to carry User
When
open, SIP URI
will carry user = phone parameter; When
closed, SIP URI does not carry user = phone parameter.
Use PSTN CID
This parameter is only for the FXO port to the PSTN get
to use the explicit when turned on,
to display the number of
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