GoIP User Manual
http://www.dbltek.com
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protocol supported is PPTP with no encryption or 40-bit encryption which is defined on the VPN server.
In general, this option is used to avoid VoIP blockings.
3.3.3Basic VoIP
The GoIP can support both SIP and H.323 VoIP protocols.
For GoIP-1, both protocol are embedded in one
firmware. User needs to select the VoIP protocol as shown below.
As more features are added, SIP and H.323 VoIP protocols are supported in two different firmware versions.
Except GoIP-1, all other models are now shipped with the SIP protocol firmware as a factory default.
If
H.323 protocol is required, the firmware of the device can be changed to the one that supports H.323
protocol.
Please visit our website for the latest firmware versions or contact us or your supplier for the
latest firmware upgrade links available.
In general, it is important to understand your VoIP application with the device before proceeding to device
configuration.
If the device is going to work with a IP PBX, please make sure that you know how to configure
your IP PBX.
It is very important that you send us your application requirements in full details when seeking
for technical support in configuring the device.
In order to simplify SIP configuration, SIP settings are categorized as Basic VoIP, Advanced VoIP and Media.
In general, Basic VoIP defines how the GoIP handle SIP calls and four SIP modes are supported.
It is
important to understand the differences between each mode in order to select a mode that is the most
suitable for your application.
Depending on the SIP environment and network conditions, you may or may
not need to change the default settings in the Advanced VoIP and Media pages.
Once SIP settings are completed, it is important to configure the device for making outgoing calls and
receiving incoming calls.
Please see section 3.3.6 and 3.3.7 for more information on Call OUT and Call IN
settings.
The four modes of SIP operations are described below.
1. Single Server Mode
In this mode, only one SIP registration is used for single or multiple-line operation.
Please make sure that
your SIP server supports multiple-line operation and the SIP account is configured in the SIP server to
match the number of lines available in the device.
Call routing to a GSM channel is now based on the
Routing Prefix of each GSM channel.
Here is the channel selection algorithm.
