Proceed as follows:

Requirements:

Select the provider to be displayed/configured.

Number of VoIP channels provided by the VoIP provider. If these channels are occupied additional connections required are made using another provider.

Number of VoIP channels provided by the VoIP provider to be used only for incoming/outgoing calls.

Requirements:

Forces disabling of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in sub-system operation or external VoIP subscriber) requires call data to be encrypted using SRTP.

Switches on negotiation for the encryption of call data by SRTP. When a call is made, the call partner will be asked if encryption is possible. If so, the call data will be encrypted and then transmitted, if not, the call data will not be encrypted.

Forces enabling of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in sub-system operation or external VoIP subscriber) does not require call data to be encrypted using SRTP.

NAT (Network Address Translation) enables you to convert a particular IP address that is used within a network (e.g. a local network) into a different IP address that can be used by a different network (e.g. the public switched phone network). If NAT traversal is switched on, and a query is sent from a local IP address to the public network, the sending IP address is swapped with the public IP address. This function is performed in the reverse direction for the reply.

Disabled (use local address): NAT traversal is not performed by the PBX. The router which connects the local area network with the Internet should be a properly functioning, SIP-aware router which performs NAT traversal.

Enabled with use of STUN: NAT traversal is performed by the PBX. To achieve this, a STUN server must also be specified for the VoIP provider.

Requirements:

NAT traversal is performed by the PBX. The VoIP provider requires a connection via a static public IP address.

Requirements:

The PBX uses the same channel to transfer the DTMF signals and the voice data (DTMF tones).

The PBX uses different channels to transfer the DTMF signals and the voice data. The DTMF tones are filtered out of the voice data.

The PBX transmits the DTMF signals on two channels (1st contains DTMF tones together with the voice data + 2nd contains the DTMF signal).

This compensates for local echoes and reverberation effects.

This function allows to add the header SilenceSupp: on/off to the SDP negotiation of the SIP packets.

If automatique was selected (factory settings), the header is not set.

The options sur or Arrêt should only tentatively be used for known problems with silent suppression.

Otherwise always select automatique.

This function allows to decide whether comfort noise is offered during the SDP media negotiation (sur) or not (Arrêt).

This function is only needed in case of compatibility problems with comfort noise. In this case comfort noise support should be switched off. Otherwise always select automatique (factory settings).

The size of the jitter buffer specifies how many RTP packets can be cached, to buffer disruptions or compensate for them.

Specifies the maximum size of a received packet for RTP supported by the PBX (30 ms recommended).

Suppresses the output of the maxptime header.

The PBX attempts to determine the required value itself.

The PBX generates the exchange line replacement tone in the case, for example, of longer dialling phases during which the PBX has not yet sent a call tone. If you find this disruptive or annoying, it can be switched off.

The PBX generates the exchange line replacement tone in the case of longer dialling phases during which there is no call tone.

No exchange line replacement tone is generated. No acoustic acknowledgement during longer dialling phases during which there is no call tone.

Requirements:

The PBX makes various codecs available. The selection of a codec affects the quality of a VoIP call. Different codecs can be configured, depending on the available connection bandwidth, ranging from codecs with the best possible VoIP call quality down to codecs with high compression (low bandwidth). The codec actually used for a call is not fixed until the codec is negotiated with the VoIP provider.

Selects a codec sequence with best possible VoIP call quality (high bandwidth) as highest priority.

Selects a compromise between VoIP call quality (high bandwidth) and compression (lower bandwidth) for the codec sequence.

Selects a codec sequence with stronger compression (lower bandwidth) as highest priority.