SIP provider line

Name Name of the line; may not be identical to another line that is configured in the device.

Mode This selection determines the operating mode of the SIP line. Possible values:

Single account mode: Externally, the line behaves like a typical SIP account with a single public number. The number is registered with the service provider, the registration is refreshed at regular intervals (when (re-)registration has been activated for this SIP provider line). For outgoing calls, the calling-line number is replaced (masked) by the registered number. Incoming calls are sent to the configured internal target number. The maximum number of simultaneous connections is either set by the provider or it depends on the available bandwidth and the codecs being used.

Table for number translation:

Single account	SIP number incoming to the line	SIP number sent from the line
Outgoing call	“From:”	The number registered at the provider (User ID)
Incoming call	“To:”	User ID

Trunk mode: Externally, the line acts like an extended SIP account with a main external telephone number and multiple extension numbers. The SIP ID is registered as the main external number with the service provider and the registration is refreshed at regular intervals (when (re-)registration has been activated for this SIP provider line). For outgoing calls, the switchboard number acts as a prefix placed in front of each calling number (sender; SIP: “From:”) . For incoming calls, the prefix is removed from the target number (SIP: “To:”). The remaining digits are used as the internal extension number. In case of error (prefix not found, target equals prefix) the call is forwarded to the internal target number as configured. The maximum number of connections at any one time is limited only by the available bandwidth.

Table for number translation:

Trunk	SIP number incoming to the line	SIP number sent from the line
Outgoing call	“From:”	Switchboard number (User-ID) + “From:”
Incoming call	Switchboard number (User-ID) + “To:”	“To:” As internal extension

Gateway mode: Externally the line behaves like a typical SIP account with a single public number, the SIP ID. The number (SIP ID) is registered with the service provider and the registration is refreshed at regular intervals (when (re-)registration has been activated for this SIP provider line). For outgoing calls, the calling-line number (sender) is replaced (masked) by the registered number (SIP ID in SIP: “From:”) and transmitted in a separate field (SIP: “Contact:”) . For incoming calls the dialed number (target) is not modified. The maximum number of connections at any one time is limited only by the available bandwidth.

Table for number translation:

Gateway	SIP number incoming to the line	SIP number sent from the line
Outgoing call	“From:”	The number registered at the provider (User ID)
	“From:”	“Contact:”
Incoming call	“To:”	“To:”

Link mode: Externally, the line behaves like a typical SIP account with a single public number (SIP ID). The number is registered with the service provider, the registration is refreshed at regular intervals (when (re-)registration has been activated for this SIP provider line). For outgoing calls, the calling-line number (sender; SIP: "From:") is not “From:”) is not modified. For incoming calls, the dialed number (target; SIP: “To:”) is not modified. The maximum number of connections at any one time is limited only by the available bandwidth.

Table for number translation:

Link	SIP number incoming to the line	SIP number sent from the line
Outgoing call	“From:”	“From:”
Incoming call	“To:”	“To:”

Domain SIP domain/realm of the upstream device. Provided the remote device supports DNS service records for SIP, this setting is sufficient to determine the proxy, outbound proxy, port and registrar automatically. This is generally the case for typical SIP provider services.

Rtg tag Routing tag for selecting a certain route in the routing table for connections to this SIP provider.

Port TCP/UDP port that the SIP provider uses as the target port for SIP packets.

Note: This port has to be activated in the firewall for the connection to work.

User ID Telephone number of the SIP account or name of the user (SIP URI).

Note: For a SIP trunking account, the switchboard number is entered here. For incoming calls, any numerals after the switchboard number are interpreted as extension numbers (DDI) and these are passed to the call router. For outgoing calls, DDI numbers received from the call router are combined with the switchboard number. This access data is used to register the line (single account, trunk, link, gateway), but not the individual local users with their individual registration details. If individual users (SIP, ISDN, analog) are to register with an upstream device using the data stored there or on the terminal device, then the line type "SIP PBX line" should be selected.

Auth-Name Name for authentication to the upstream SIP device (provider/SIP PBX).

Note: This access data is used to register the line (single account, trunk, link, gateway), but not the individual local users with their individual registration details. If individual users (SIP, ISDN, analog) are to register with an upstream device using the data stored there or on the terminal device, then the line type "SIP PBX line" should be selected.

Display name Name for display on the telephone being called.

Note: Normally this value should not be set as incoming calls have a display name set by the SIP provider, and outgoing calls are set with the local client or call source (which may be overwritten by the user settings for display name, if applicable). This settings is often used to transmit additional information (such as the original calling number when calls are forwarded) that may be useful for the person called. In the case of single-line SIP accounts, some providers require an entry that is identical to the display name defined in the registration details, or the SIP ID (e.g. T-Online). This access data is used to register the line (single account, trunk, link, gateway), but not the individual local users with their individual registration details. If individual users (SIP, ISDN, analog) are to register with an upstream device using the data stored there or on the terminal device, then the line type "SIP PBX line" should be selected.

Secret The password for authentication at the SIP registrar and SIP proxy at the provider. For lines without (re-)registration, the password may be omitted under certain circumstances.

Note: This access data is used to register the line (single account, trunk, link, gateway), but not the individual local users with their individual registration details. If individual users (SIP, ISDN, analog) are to register with an upstream device using the data stored there or on the terminal device, then the line type "SIP PBX line" should be selected.

Registrar The SIP registrar is the point at the SIP provider that accepts the login with the authentication data for this account.

Note: This field can remain empty unless the SIP provider specifies otherwise. The registrar is then determined by sending DNS SRV requests to the configured SIP domain/realm (this is often not the case for SIP services in a corporate network/VPN, i.e. the value must be explicitly set).

Outb-proxy The SIP provider's outbound proxy accepts all SIP signaling originating from the LANCOM device for the duration of the connection.

Note: This field can remain empty unless the SIP provider specifies otherwise. The outbound proxy is then determined by sending DNS SRV requests to the configured SIP domain/realm (this is often not the case for SIP services in a corporate network/VPN, i.e. the value must be explicitly set).

Cln prefix The call prefix is a number placed in front of the caller number (CLI; SIP "From:") for all incoming calls on this SIP provider line in order to generate unique telephone numbers for return calls. For example; a number can be added, which the call router analyzes (and subsequently removes) to select the line to be used for the return call.

Number/Name The effect of this field depends upon the mode set for the line:

If the line is set to "Single account" mode, all incoming calls on this line with this number as the target (SIP: "To:") and transferred to the call router.
If the mode is set to "Trunk", the target number is determined by removing the trunk's switchboard number. If an error occurs, the call will be supplemented with the number entered in this field (SIP: "To:") and transferred to the call router.
If mode is set to "Gateway" or "Link" the value entered in this field has no effect.

Codecs While the connection is being established, the terminal equipment negotiates the codecs that are to be used for voice-data compression. Use the codec filter to restrict the codecs that are permitted and to permit only certain codecs.

Note: If no common the codecs can be agreed upon, no connection is made.

Codec order This parameter influences the order in which the codecs are presented during connection establishment.

Refer Call switching (connect call) between two remote subscribers can be handled by the device itself (media proxy) or it can be passed on to the exchange at the provider if both subscribers can be reached on this SIP provider line (otherwise the media proxy in the LANCOM device assumes responsibility for switching the media streams, for example when connecting between two SIP providers).

Note: An overview of the main SIP providers supporting this function is available in the Support area of our Internet site.

Local port number This is the port used by the LANCOM proxy to communicate with the provider.

Note: If line (re-)registration is deactivated, the local port has to be defined with a fixed value, and this also has to be entered at the provider end as the destination port (e.g. when using an unregistered trunk in the company VPN). This ensures that both ends can send SIP signaling.

(Re-) registration This activates the (repeated) registration of the SIP provider line. Registration can also be used for line monitoring.

Note: To use (re-) registration, the line monitoring method must correspondingly be set to "Register" or "Automatic". Registration is repeated after the monitoring interval has expired. If the provider's SIP registrar suggests a different interval, the suggested value is used automatically.

Line monitoring Specifies the line monitoring method. Line monitoring checks if a SIP provider line is available. The Call Router can make use of the monitoring status to initiate a change to a backup line. The monitoring method sets the way in which the status is checked. Possible values:

Auto: The method is set automatically.
Disabled: No monitoring; the line is always reported as being available. This setting does not allow the actual line availability to be monitored.
Register: Monitoring by means of register requests during the registration process. This setting also requires "(Re-)registration" to be activated for this line.
Options: Monitoring via Options Requests. This involves regular polling of the remote station. Depending on the response the line is considered to be available or unavailable. This setting is well suited for e. g. lines without registration.

Monitoring interval The monitoring interval in seconds. This value affects the line monitoring with register request and also the option request. The monitoring interval must be set to at least 60 seconds. This defines the time period that passes before the monitoring method is used again. If (re-) registration is activated, the monitoring interval is also used as the time interval before the next registration.

Note: If the remote station responds to an option request with a different suggested value for the monitoring interval, this is accepted and subsequently applied.

Trusted Specifies the remote station on this line (provider) as "Trusted Area". In this trusted area, the caller ID is not concealed from the caller, even if this is requested by the settings on the line (CLIR) or in the device. In the event of a connection over a trusted line, the Caller ID is first transmitted in accordance with the selected privacy policy and is only removed in the final exchange before the remote subscriber. This means, for example, that Caller ID can be used for billing purposes within the trusted area. This function is interesting for providers using a VoIP router to extend their own managed networks all the way to the connection for the VoIP equipment.

Note: Please note that not all providers support this function.

Privacy method Specifies the method used for transmitting the caller ID in the separate SIP-header field.

Active Activates or deactivates the entry.

Comment Comment on this entry.