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adVanced gateWay settIngs
dIal PrefIx
Some Gateways require a certain prefix be dialed in order to select a particular legacy port for outgoing calls. If this
is required for your Gateway (e.g. 991) enter it here.
outgoIng enabled
Allow outgoing calls on channels using this Gateway using the VoIP companion phone.
sIP trunks
With regard to STAC VIP, we refer to SIP Trunks as provider accounts that don’t require registration with a provider’s
server. These types of services are also referred to as using “IP Authorization” because call authorization is
accomplished by sending calls to a specific, fixed IP address at the customer’s site. Likewise, outgoing calls are
authorized if they come from the customer’s static IP address.
In this way, SIP Trunks are simpler on the provider side because there is no username or password associated
with the account, and no login procedure. Trunks can be more complex on the STAC VIP side, because STAC VIP
must have a way to filter incoming calls as being from your provider, and not some random call on your network.
If setting up a SIP trunk behind a router or firewall, special port forwarding rules will need to be applied to your
network. Your trunking provider often has guides for this process.
You may also come across certain PBXs that must deliver their extensions in this way. In order for STAC VIP to
receive these extensions, a SIP trunk must be created instead of a provider account.
SIP Trunks also differ from normal SIP providers in several ways. Rather than having STAC VIP “pull” the SIP channel
from a provider, a SIP trunking provider will “push” the channel to a specific IP address of the user. This means in
order to support SIP trunking you need a public, static IP address, and no other devices can be utilizing the SIP ports
at that address.
Unlike normal SIP providers,
only a single SIP Trunk is supported on each STAC VIP mainframe
.
The option to add a new SIP Trunk is located in
Line Configuration-> VoIP providers-> Add Provider -> SIP Trunk
.
Once a new Trunk is created, there will be no option to create another.
Although the settings menu for SIP trunking appears the same as for a normal provider, only a few of them are
meaningful. You should put your trunking provider’s name into the “
name
” field, choose your codec options (see
normal provider settings) and set the correct SIP port value (usually
5060
).
Username
and
Password
fields can be
ignored.
You’ll also need to populate the
Server/Realm
field in the trunk settings menu. This will be the IP address (e.g.
74.94.151.151
) of the Trunking provider. Unlike with registered providers, this field should not be populated with
Summary of Contents for VH2
Page 1: ...Product Manual ...
