SIP lines

The device uses these lines to register with other SIP remote stations (usually SIP providers or remote gateways at SIP PBXs). The connection is made either over the Internet or a VPN tunnel.

The settings are configured under Voice Call Manager > Lines by clicking the button SIP lines.

The General tab contains the following configuration options:

Entry active

Activates or deactivates this entry.

Mode

This selection specifies the operating mode of the SIP line. Possible values are:

Single account

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 (if (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 destination 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.

Trunk

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 switchboard number with the service provider and the registration is refreshed at regular intervals (if (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 destination number (SIP: “To:”). The remaining digits are used as the internal extension number. In case of error (prefix not found, destination equals prefix) the call is forwarded to the internal destination number as configured. The maximum number of connections at any one time is limited only by the available bandwidth and possibly by the provider.

Gateway

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 (if (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 sent in a separate field (SIP: “Contact:”). For incoming calls the dialed number (destination) is not modified. The maximum number of connections at any one time is limited only by the available bandwidth and possibly by the provider.

Link

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 (if (re-)registration has been activated for this SIP provider line). For outgoing calls, the calling-line number (sender; SIP: "From:") is not modified. For incoming calls, the dialed number (destination; SIP: "To:") is not modified. The maximum number of connections at any one time is limited only by the available bandwidth and possibly by the provider.

Flex

To the outside the line behaves like a commercially available SIP account with a single public number.
The number is registered at the service provider and registration is refreshed on a regular basis.
For outgoing calls, the calling-line number (sender) is not modified.
For incoming calls the dialed number (destination) is not modified.
The maximum number of connections at any one time is limited only by the available bandwidth.

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

Name

The name of the line: This may not be the same as another line (SIP provider, ISDN or SIP PBX) configured on the device.

Comment

Comment on this entry.

SIP domain/realm

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.

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).

Note: To force SIP registration via IPv4 or IPv6 the SIP domain can be entered in the field Registrar followed by the suffix ?4 or ?6 (e.g. SIP-Domain.com?4). See also Configuration option for IPv4/IPv6 resolution with DNS resolutions.

Outbound proxy

The SIP provider's outbound proxy accepts all SIP-call signaling that originates from the device for the duration of the connection.

Note: This field can remain empty unless the SIP provider specifies otherwise. In this case, the outbound proxy is identical to the registrar. This is a typical configuration for SIP-provider offerings.

Port

This is the remote port used to communicate with the provider.

Switching at provider active

Call switching (transfer 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. The advantage of this is that the LANCOM VoIP router no longer requires the bandwidth. Otherwise, the media proxy in the device switches the media flows, such as when connecting two SIP provider lines.

Note: Switching at the provider will only work if both connections are routed via the same provider line.

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

(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, set the line monitoring method on the Advanced tab to "Register" or "Automatic". The device renews its registration after the monitoring interval expires. If the provider's SIP registrar suggests a different interval, the devices uses this value automatically.

SIP-ID/user

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 either there or on the terminal device, then a SIP-PBX line should be set up.

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 either there or on the terminal device, then a SIP-PBX line should be set up.

Authentication 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 either there or on the terminal device, then a SIP-PBX line should be set up.

Password

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.

Call prefix

The device places a call-prefix number in front of the caller number (CLI; SIP "From:") for all incoming calls on this SIP line. This generates unique telephone numbers for return calls. For example; you add a number here, which the call router analyzes (and subsequently removes) to select the line to be used for the return call.

Internal destination number

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 destination (SIP: "To:") are transferred to the call router.
If the mode is set to "Trunk", the destination 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:") are transferred to the call router.
If mode is set to "Gateway" or "Link" the value entered in this field has no effect.

The Security tab contains the following configuration options:

Signaling encryption

This setting determines the protocol used for signaling encryption (SIP/SIPS) for communications with the provider.

Automatic: NAPTR (Naming Address Pointer) records are used for DNS resolution. In the DNS data, the provider specifies the use of transport protocols such as UDP, TCP or TLS. The provider can also specify weights or priorities. If TLS is specified as the transport protocol for signaling encryption by NAPTR, voice encryption is also used automatically, regardless of the explicit configuration setting of voice encryption.
No (UDP): All SIP packets are transmitted connectionless. Most providers support this setting.
No (TCP): All SIP packets are transmitted connection-oriented. The device establishes a TCP connection to the provider and maintains it for as long as it stays registered. Specialized providers, such as the providers of SIP trunks, support or force this setting.
TLS: Transmission is the same as with TCP, but all of the SIP packets are encrypted all the way to the provider. The TLS version selected in the configuration is taken as the minimum requirement for TLS encryption.

Speech encryption

This setting determines if and how the speech data (RTP/SRTP) is encrypted when communicating with the provider.

Speech encryption
Reject	Encryption is not available for outgoing calls. Incoming calls with an encryption proposal are rejected. The speech channel is not encrypted.
Ignore	Encryption is not available for outgoing calls. Incoming calls with an encryption proposal are accepted. The speech channel is not encrypted.
Prefer	Encryption is offered for outgoing calls. Incoming calls without an encryption proposal are accepted. The speech channel is only encrypted if the remote peer also supports encryption.
Force	Encryption is offered for outgoing calls. Incoming calls without an encryption proposal are rejected. The speech channel is either encrypted or is not established.

Note: If you require the encrypted transmission of speech data, the signaling must also use an encrypted channel. Please note that the use of SRTP is no guarantee of end-to-end encryption.

Verify server cert. acc. to:

With this setting, you specify whether the certificate of the SIP server is verified against certain Certificate Authorities (CAs). CA certificates from globally recognized certificate chains are updated with LCOS updates. They can also be manually updated by truststore updates.

Server certificate
No verification	The server certificate is not verified. All valid server certificates are accepted, whichever CA they were signed by. This setting is useful for accepting self-signed certificates.
All trusted CAs	The server certificate is verified against all CAs known to the device. These include all CAs that LCOS "knows" to be trusted and also those from the VoIP server certificate slots 1 to 3. Note: The encrypted connection is only established if one of these certificates is validated successfully.
VoIP cert. slot 1	A check is made to see whether the server certificate was signed by the CA whose certificate was uploaded to slot 1 of the VoIP certificates.
VoIP cert. slot 2	A check is made to see whether the server certificate was signed by the CA whose certificate was uploaded to slot 2 of the VoIP certificates.
VoIP cert. slot 3	A check is made to see whether the server certificate was signed by the CA whose certificate was uploaded to slot 3 of the VoIP certificates.
Telekom-Shared-Business-CA4	With this setting, the device only accepts server certificates signed by the Telekom Shared Business CA4 CA. Note: Use this setting for SIP trunk connections from Deutsche Telekom.

Fallback SIPS > SIP

No: No fallback to an unencrypted connection is performed. If it is not possible to establish an encrypted connection to the VoIP provider, the line remains unregistered.
UDP: As a rule, encrypted SIP connections are made with the TCP protocol and unencrypted connections are made with the UDP protocol. This setting switches directly to an unencrypted UDP connection if the encrypted TCP connection cannot be established.
Complete: If an encrypted TCP connection with the configured TLS version cannot be established, then attempts are made to establish an unencrypted TCP connection, and finally a UDP connection in order to register the VoIP line.
Note: This setting provides the best compatibility, but may lead to a longer registration time.

Allow inbound UDP packets

If the provider line uses UDP to communicate with the registrar, it receives UDP packets on the desired local port. With this setting you specify the network context in which a UDP packet is accepted. The device only accepts a packet from the WAN / VPN / LAN if you have activated the corresponding setting. Otherwise the packet is dropped.

Allow SIP messages only from registrar (strict mode)

If this mode is activated, incoming SIP messages are only accepted from IP addresses that were reported by the provider when the domain / registrar was resolved. If the VoIP provider signals a call from an IP address that was not included in the DNS resolution of the domain / registrar, the incoming call is not signaled to the internal subscriber.

Important: Deactivating this function increases the risk of cyber attacks. SIP messages sent by an attacker can lead to calls being established and unwanted costs. SIP messages that are forwarded to internal clients can potentially exploit security vulnerabilities in the terminal devices.

On the tab Advanced you configure the SIP proxy, the line monitoring, and the calling line identification restriction.

SIP proxy port

This is the local port used by the SIP-proxy device to communicate with the remote station. By default "0" is set here. The port is dynamically selected from the pool of available port numbers. You can also specify of a port in the range of "1" to "65535".

Routing tag

This routing tag selects a certain route in the routing table for connections to this SIP server.

Source address

The device automatically determines the correct source IP address for the destination network. To use a fixed source IP address instead, enter it symbolically or directly here.

Control method

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 are:

Automatic: The method is set automatically (default).
Deactivated: No monitoring. The line is reported as available when the option (re)registration is disabled. Otherwise it will be considered to be available only after a successful registration. 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 example, for lines without registration.

Monitoring interval

The monitoring interval in seconds. This value affects the line monitoring with 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.

Trusted area activated

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.

Transmission method

Specifies the method used for transmitting the caller ID in the separate SIP-header field. Possible values are:

None: The default setting, so no transmission takes place.
RFC3325: Transmission according to "P-Preferred-Id/P-Asserted-Id".
IETF-Draft-Sip-Privacy-04: Transmission according to "IETF-Draft-Sip-Privacy-04" by means of RPID (Remote Party ID).

DTMF signaling

Depending on the requirements, it may not be sufficient to transmit "inband" DTMF tones if a SIP receiver cannot recognize these. In this case, it is possible to configure an alternative method of DTMF transmission for All-IP connections.

Only in-band (in audio): The tones are transmitted as DTMF tones (G.711) in the RTP (voice) stream.
Only SIP info: The DTMF tones are transmitted "out-of-band" as a SIP-info message with the parameters Signal and Duration (as per RFC 2976). There is no parallel transmission of G.711 tones.
Telephone events – fallback to in-band (default): The DTMF tones are transmitted as specially marked events within the RTP stream (as per RFC 4733). There is no parallel transmission of G.711 tones. If the call-initialization SDP message does not include telephone-event signaling, negotiations fallback to inband transfer as per G.711.
Telephone events – fallback to SIP info: The DTMF tones are transmitted as specially marked events within the RTP stream (as per RFC 4733). There is no parallel transmission of G.711 tones. If the call-initialization SDP message does not include telephone-event signaling, negotiations fallback to transfer as per SIP-Info message.

Overlap dialing

Overlap dialing significantly reduces the waiting time between the number being dialed and the call being established. With overlap dialing disabled, your LANCOM device uses an overlap timer. The factory setting for this is 6 seconds. If the timer expires without you dialing any further numbers, the number entered so far is considered to be complete and the call is established. With overlap dialing enabled on the line, portions of the dialed number are immediately sent to the All-IP provider. If the All-IP provider responds with "484 number incomplete", the Voice Call Manager collects any additional dialed digits and sends them to the exchange again. In this way, calls are established as quickly as possible without the 6 second delay, as you are accustomed to from your ISDN connection.

Note: However, since this functionality is not supported by all SIP providers, overlap dialing has to be configured for each individual SIP line.

Call forwarding using SIP 302

Activates call forwarding via SIP 302 at the SIP provider. See also Call forwarding (call deflection / partial rerouting) at the SIP trunk (SIP 302).

SIP-ID transmission

This field sets the way in which the SIP ID is transmitted for outgoing calls when operating a SIP trunk. Depending on the provider, it may be necessary to transmit the SIP ID via a different field, as otherwise the call might be rejected by the provider. The following values can be selected:

P-Preferred-Identity (default value)
FROM
None
P-Preferred-Identity without DDI
PPI-PPI
None – PPI (P-Preferred-Identity)
None – PAI (P-Asserted-Identity)

Selecting the option P-Preferred-Identity (PAI- PPI) transmits the SIP ID including the DDI via the PPI/ PAI. The source telephone number is transmitted via the FROM field. Selecting the option FROM transmits the SIP ID via the FROM field. The source telephone number is transmitted via the PPI / PAI field. With the setting None, the SIP ID is not transmitted. The first calling number is transmitted with FROM, the second in the PPI / PAI. In contrast to the P-Preferred-Identity, the setting P-Preferred-Identity without DDI does not transmit an extension number (DDI) in the SIP ID via the PPI. Selecting the option PPI- PPI (PPI) transmits the SIP ID including the DDI via the PPI. The source telephone number is transmitted via the FROM field. With the setting None – PPI (P-Preferred-Identity, the SIP ID is not transmitted. The first calling number is transmitted with FROM, the second in the PPI. With the setting None – PAI (P-Asserted-Identity, the SIP ID is not transmitted. The first calling number is transmitted with FROM, the second in the PAI.

Note: With a single account, outgoing calls always signal the SIP ID in the FROM field.