Label
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Explanation
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Name
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Name of the Trunk Group
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Description
|
Description of the Trunk Group
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Type
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Type of Trunk Group
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Profile
|
Trunk Group Profile
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Location
|
Location of Trunk Group
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Calling Rights
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Defines where incoming calls of this trunk group may be forwarded to if the call destination is not a user at the same SwyxServer
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Label
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Explanation
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Calling Rights
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Select where incoming calls received via this trunk group may be forwarded to. This allows you to specify whether and which other trunk groups the call may use to leave this SwyxWare-Installation if its destination is not a user of this SwyxServer:
Internal connections only (default value)
International connections
Calls within Europe
National connections
local calls
Deny all calls
If a user has been called on this and the call is forwarded by his Call Routing, the call gets the permissions of the called user.
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Selection prefix for the trunk group
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Set a prefix that allows a user to route the call specifically through this trunk group.
The selection prefix must be uniquely assigned to a trunk group, it cannot be assigned more than once, it may only consist of the characters '01234567890#*' and may not begin with '##'.
Examples:
In the following, the project code *1234# and the selection prefix **34#
<*Project code#><Selection prefix><Canonical call number>
*1234#**34#+44123555777
or if using a public line access
<*Project code#><Trunk Groups prefix><Public line access><National number>
*1234#**34#00123555777
or if using an internal number
<*Project code#><Trunk Groups prefix><Internal number>
*1234#**34#123555777
or if using an SIP URI (always beginning with sip:)
<*Project code#><Trunk Groups prefix><SIP:URI>
*1234#**34#sip:han.solo@millenium-falcon.com
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Public line access of the superior PBX
|
If SwyxWare is configured as a sub-PBX, enter the outside line access of the superordinate PABX.
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Label
|
Explanation
|
---|---|
Conversion for outgoing calls
|
Procedures for converting outgoing numbers and interpreting incoming numbers are defined within a trunk group. In the properties of a trunk group, the selected profile is used to specify in detail which phone number (outgoing or incoming and calling or called number) is converted into which format. This mapping of formats can be modified subsequently by administrators.
For the available phone number formats, see section "Supplied configuration files" under
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Conversion for incoming calls when number type is unknown
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Label
|
Explanation
|
---|---|
STUN support
|
Check the box to enable STUN support.
STUN can be used to determine the current public IP address of the connection so that the remote station can address and return its call data correctly.
|
STUN server
|
If your SIP provider supports STUN, enter the name or IP address of your provider's STUN server and the corresponding port.
Alternatively, you can use the free STUN server "stunserver.org" with port "3478".
|
STUN server port
|
|
Outbound proxy
(SIP only)
|
Some providers have an outbound proxy before the SIP proxy. If necessary, enter these parameters according to your provider's specifications.
|
Outbound proxy port
(SIP only)
|
|
Proxy
(SIP only)
|
Enter the address and port of the proxy server.
The SIP proxy server takes over the connection setup to the appropriate subscriber, first checking which SIP registrar the relevant subscriber is logged in with. Upon request, the subscriber then receives the current IP address of the subscriber and can thus deliver the call to this address.
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Proxy port
(SIP only)
|
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Realm
(SIP only)
|
If necessary, enter the realm area of the provider.
An SIP URI (userId@realm) is derived from the user ID (userId), the configuration of the SIP account, and the realm of the provider (realm). If not specified, the Registrar or Proxy value is used.
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DTMF method:
(SIP only)
|
Select a DTMF method if necessary.
This mode is used to specify how the provider handles keystrokes from the user (DTMF signaling).
None.
DTMF signalling is deactivated.
RFC2833_Event:
DTMF signaling is used based on the event mechanism described in RFC2833.
Info Method DTMF Relay
DTMF signaling is used as suggested by Cisco (applicationtype DTMFRelay).
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Label
|
Explanation
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---|---|
Transport protocol
|
Select the transport protocol to be assigned to the trunk group:
Automatic (Standard)
The transport protocol is determined automatically by DNS lookup.
UDP
This transport protocol is supported by most SIP providers. It requires the lowest bandwidth, however, it carries a higher risk of data loss.
TCP
This transport protocol is known to be reliable, however, it requires higher bandwidths.
TLS
This transport protocol has TCP characteristics and supports encryption. When selecting this protocol SIP packet are transmitted encrypted.
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Encryption mode
|
Select the encryption mode.
This setting is enabled only if you have selected the "TLS" transport protocol. You can define if voice data will also be encrypted when using the secure TLS connection.
No encryption
The voice data is not encrypted.
Encryption mandatory
The voice data is encrypted between SIP provider and SwyxLinkManager.
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