Label | Explanation |
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Name | Enter a name. The name must be unambiguous within SwyxWare. |
Description | Enter a description if necessary. |
Trunk Group | Select a trunk group of the required type to which the trunk is to be assigned: The trunk assigned to this trunk group gets the property from the corresponding type: |
SIP trunk SIP trunks enable the use of VoIP services. The service provider usually assigns a number range or SIP URIs. If the service provider in question also offers gateway services, it is also possible to reach numbers in the public telephone network via a SIP trunk and the provider's gateway behind it. | |
SIP gateway trunk SIP gateway trunks are used to control gateways that are themselves reached by SwyxServer via a SIP connection. This allows, for example, telephones in small branches and branch offices to be operated with a local gateway in each case and with a local direct connection to the public telephone network. Only gateways for which profiles are included in the delivery are supported at present. | |
ENUM trunk An ENUM link enables you to make SIP calls with ENUM number resolution via the Internet. This means, for example, that the user of a SIP telephone can automatically determine the SIP address of the desired call partner simply by entering the telephone number and have the telephone number converted to the SIP address. The called party can then be reached over the IP network in spite of using a 'normal' phone number. This requires that the desired call partner is registered with ENUM. |
Label | Explanation |
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SIP User ID | Enter here the SIP account data with which the SIP gateway should log on to SwyxServer via this trunk. This logon data must be entered in the same format as when was configured. SIP User ID is the user ID that together with the realm forms the SIP address (URI). |
Authentication method | Select whether the gateway should authenticate. |
User name | The user name and password are required for user authentication. |
Password | |
Repeat password |
Label | Explanation |
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SIP Provider | SIP Provider Profile. This property is inherited from the assigned trunk group. |
SIP User ID | Enter the user data you received from your SIP provider: SIP User ID is the user ID that together with the realm forms the SIP address (URI). |
SIP user name | The user name and password are required for user authentication. |
Password | |
Repeat password |
Country and area codes are specified by the location of the trunk group. |
Label | Explanation |
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Country code | If necessary, enter the country code. e. g. 49 (for Germany) |
Area Code | If necessary, enter the area code. e. g. 30 (for Berlin) |
First subscriber number | Enter the public phone numbers to be used by this trunk. External calls to these numbers go over this trunk. Calls with a calling party number assigned to this trunk are routed via this trunk. If you get several individual phone numbers or several phone number ranges set up by your provider, specify only one range and add the others later, see Edit Trunks |
Last competitor number |
Label | Explanation |
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Username | If necessary, enter the SIP addresses (URIs) that this trunk should manage. A SIP has the following format: SIP:<Username>@<Realm> For simplification, you can use '*' as a placeholder here, e.g. '*@company.com' represents all users with the realm 'company.com'. |
Realm | The realm is already specified by the selection of the trunk group, but can be overwritten. |
Label | Explanation |
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Codec priority and filter | Select the type of compression to be used on this trunk: Prefer quality The codecs are provided in the order G.722, G.711a, G.711µ, G.729, Fax over IP. Prefer low bandwidth The codecs are provided in the order G.729, G.722, G.711a, G.711µ, Fax over IP. It is important to use as little bandwidth as possible. You can disable unwanted codecs: |
G.711µ (approx. 64 kBit/s per call) | Voice, high bandwidth (G.711a, G.711µ) The voice data is slightly compressed. This keeps the packet delay time in the LAN (Local Area Network) to a minimum. |
G.711a (approx. 64 kBit/s per call) | |
G.722 (around 84 kbit/s per call) | Voice, highest bandwidth (G.722) HD quality |
G.729 (around 24 kbit/s per call) | Language, low bandwidth. High compression. |
Fax over IP (T.38, around 20 kbit/s per call) | The special fax protocol T.38 takes into account the conditions of an IP network. |
Label | Explanation |
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Number of channels | If necessary, enter how many calls may be made simultaneously via this trunk. Basically, the maximum number of channels depends on the available bandwidth, as well as the codec setting, i.e. the bandwidth per call. With a SIP trunk, the provider determines the maximum number of simultaneous connections that are possible. |
Label | Explanation |
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Computer name | Apply the default computer name. |