Label | Explanation |
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Name | Enter the name of the Trunk Group. The name must be unambiguous within SwyxWare. |
Description | If applicable, enter a brief description. |
Location | Select a defined Location to which the new Trunk Group should be assigned. |
Type | Select the Type of the Trunk Group: ISDN SIP SIP Gateway |
Profile | Select the Trunk Group profile. A trunk group profile specifies how the trunk interprets and handles the call numbers. Depending on the trunk type, a number of predefined profiles are available. For each of these profiles, the number format is specified. For SIP trunks in particular, the profile specifies the provider and the necessary SIP parameters. Please contact your specialist dealer for an overview of compatible SIP providers and the functions they offer in relation to SwyxWare. The Swyx knowledge base contains a related entry, which is constantly updated. Simply visit: |
Calling Right | Specify what rights a call has when coming in via this trunk. This determines whether and which other trunk groups it may use to leave this SwyxServer installation, provided that its destination is not a user of this SwyxServer. deny all calls internal calls Local calls National destinations European destinations no call restrictions |
Enable Trunk Recording | If you activate the "Record calls via trunks" checkbox, all calls made via this trunk group will be recorded in the way that is activated in the settings of the SwyxServer. |
Label | Explanation |
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Enable SIP registration | Activate the check box to permit SIP registration. |
Registrar | Enter the registrar. REGISTER messages are sent to this address. If no value is entered here, the value configured under proxy will be applied. |
Registrar Port | In the “Port” field you define the port on which the configured registrar receives the registration request. |
Re-registration Interval | Enter a value in seconds. The "Re-Registration Interval" defines how often the registration must be updated. A small value will allow you to quickly recognize the loss of the SIP connection to the provider. A high value results in lower network burden in standby. |
Enable STUN support | Activate the check box to permit STUN support. STUN is a network protocol that recognizes the existence and type of firewalls and NAT routers and takes this information into consideration. It enables the uncomplicated use of devices (e. g. SIP telephones) and programs in networks that should receive information from the internet. |
STUN Server | Specify the STUN server and the associated port for use by the SIP devices. |
STUN Server Port | |
Outbound Proxy | Some providers require an outbound proxy and a port. You should have received this data from your provider. |
Outbound Proxy Port | |
Proxy | Enter the SIP proxy and the port for outgoing calls. |
Port | The SIP proxy server takes over the connection setup to the appropriate subscriber, first checking which SIP registrar the relevant subscriber is logged on with. From this it requests and receives the current IP address of the subscriber, and can thus deliver the call to this address. |
Realm | Define the SIP realm 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 this field is left blank, the value registrar or proxy will be used. |
DTMF mode | Select the DTMF method. This mode defines how the provider proceeds with the keyboard input of a user during a call (DTMF signalling). None DTMF signalization is deactivated RFX2833 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 (application type DTMFRelay) will be used. |
Transport Protocol | Select the transport protocol that you want to assign the Trunk Group: Automatic 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 packets are transmitted encrypted. |
Encryption | This option will only make sense if you select TLS as the transport protocol. Activate the check box to make sure that the voice data is encrypted between SIP provider and SwyxLinkManager. |
Label | Explanation |
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Conversion for inbound calls: Calling party number | Select a number formatting profile Canonical with plus Canonical without plus CLIP no screening Dial as a PBX user Extension Fixed Subscriber ISDN Italy ISDN Netherlands CLI National Subscriber Number Transparent Type and Plan For information about the number formatting formats please refer to Number formats. |
Conversion for inbound calls: Called Party Number | |
Conversion for outbound calls: Calling party number | |
Conversion for outbound calls: Called Party Number |