Yeastar Technology TG200 User Manual page 69

Tg series gsm/3g voip gateway
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Configure SIP General Settings
Navigate to Gateway > VoIP Settings > SIP Settings > General, to configure the SIP general
settings.
Check the description of SIP general settings below.
Option
UDP Port
TCP Port
TLS Port
TLS Verify Server
TLS Verify Client
TLS Ignore Common Name
TLS Client Method
RTP Port Start
RTP Port End
DTMF Mode
Max Registration/Subscription
Time
Min Registration/Subscription
Time
Default Incoming/Outgoing
Registration Time
Register Attempts
Register Timeout
Calling Channel Codec Priority
DNS SRV Look Up
User Agent
Description
Port used for SIP registrations, the default is 5060.
Port used for SIP registrations, the default is 5060.
Port used for SIP registrations, the default is 5061.
When using TG gateway as a TLS client, whether or not to verify
server's certificate. It is "No" by default.
When using TG gateway as a TLS server, whether or not to verify
client's certificate. It is "No" by default.
Set this parameter as "No", then common name must be the same
with IP or domain name.
When using TG gateway as TLS client, specify the protocol for
outbound TLS connections. You can select it as tlsv1, sslv2 or
sslv3.
Beginning of RTP port range
End of RTP port range
Set default mode for sending DTMF. Default setting: rfc2833
Maximum duration (in seconds) of a SIP registration. The default is
3600 seconds.
Minimum duration (in seconds) of a SIP registration. The default is
60 seconds.
The default duration (in seconds) of incoming/outgoing registration.
The number of SIP REGISTER messages to send to a SIP
Registrar before giving up. Default is 0 (no limit)
Number of seconds to wait for a response from a SIP Registrar
before timed out. The default is 20 seconds.
Once enabled, when dialing out via SIP/SPS trunks, the codec of
calling channel will be selected in preference. If not, TG gateway
will follow the priority in your SIP/SPS trunks.
Please enable this option when your SIP trunk contains more than
one IP address.
To change the user agent parameter of asterisk . The default is "TG
gateway", you can change it if needed. The User Agent will show in
SIP packets
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