16.3 Creating a SIP Trunk Group
We recommend creating a SIP trunk group before creating a SIP trunk.
General parameters such as permissions, location and routings are specified in this group. When creating a trunk, you then simply assign the trunk group to the trunk. As a member of the group, the trunk is thus given the corresponding parameters.
How to create a SIP trunk group
1 Open the SwyxWare Administration and choose the SwyxServer.
2 In the left side of the SwyxWare Administration window, click with the right mouse button on "Trunk Groups" and select the entry "Add Trunk Group…" in the context menu.
The "Add Trunk Group…" wizard will appear.
3 Click on "Next>".
4 Name and description of the trunk group:
Enter the name of the trunk group, and a description.
Click on "Next>".
5 Type of trunk group:
Enter the type of trunk group here, in this case "SIP".
6 Enter the profile for this trunk group in the lower field "Profile". If your provider does not appear in the list, select "Customized".
Click on "Next >".
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If you select a predefined profile, the corresponding provider parameters will be set automatically. Configuration steps (7) "SIP settings (registration)" to (10) "Settings for the STUN server" are redundant. |
7 SIP settings (registration)
You can obtain the parameters required for the SIP connection from your SIP provider. Proceed as follows to enter them.
If the SIP provider only permits registered connections, check the "Activate SIP registration" box and enter the name of the registrar or the IP address, along with the mapped port. The registrar is the address to which the REGISTER messages are sent.
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The port must match the selected transport protocol. Leave the filed empty if you did not receive information on the port by your provider. The port is determined via DNS query.
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Enter the time after which registration must be repeated (usually two minutes) under "Re-registration interval".
Click on "Next>".
8 SIP settings
Your SIP provider will provide the following parameters:
Proxy:
Enter the name or IP address of the SIP proxy and, if applicable, the mapped port. Here, the proxy is the provider's server, which sets up and terminates the connection.
Realm:
Enter the general component of the SIP address here. This is used to create your individual SIP-URI.
Example:
Your SIP provider is outlook.com. In this case, enter "outlook.com" as the realm here. Your SIP-URI will then comprise the individual component, e.g., "tom.jones" and the realm. Your SIP-URI will, therefore, be:
tom.jones@outlook.com
DTMF method:
This mode defines how the provider proceed with the keyboard input of a user during a call (DTMF signaling).
You can choose from a variety of options:
None. DTMF signalization is deactivated
RFC2833_Event: RFC2833_Event: DTMF signalization, based on the event mechanism described in RFC2833, will be used.
Info Method DTMF Relay DTMF signalling as recommended by Cisco (applicationtype DtmfRelay) will be used
Click on "Next>".
9 Transport protocols and encryption
Select a transport protocol and, if applicable, the encryption mode.
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Make sure that the selected transport protocol is supported by your provider.
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10 Settings for the STUN server
A STUN server can be used to identify the IP address of the line. Supports your provider STUN, so please enter the name or the IP address of the STUN server of your provider and the appropriate port.
If you would like to use STUN, although you have not received any STUN server information from the provider, you can use the free STUN server "stunserver.org" with port "3478".
Click on "Next>".
11 Definition of routing:
Specify for which calls this Trunk Group should be used. When entering call numbers or URIs you can use placeholders (*), e.g. "+*" for all external numbers or "*" for all internal numbers. Multiple numbers/URIs are separated by a semicolon. You have several different options:
for all external calls
only for external calls to the following destination number or SIP-URI
for all external calls and all unassigned internal numbers
For the following internal numbers
Create no routing records for the moment
Click on "Next>".
12 Call Permission
Specify the Calling Rights profile for the Trunk Group. This Calling Rights profile applies to the incoming calls over this Trunk Group.
For further information please refer to sectionCall permission of a trunk group
Click on "Next >".
13 Location profile:
Define the location. This profile also includes the definition of e.g. country code and public line access.
Click on "Next>".
14 Click "Finish".
The new SIP trunk group is created, and is available for further configuration.
