Select the provider to be displayed/configured.

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

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

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

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

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

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

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.

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

NAT traversal is performed by the PBX. The use of STUN or rport depends on the selection for SIP NAT traversal.

If the provider requires it: NAT traversal is performed by the PBX via a static public IP address.

The PBX uses the same channel to transfer the DTMF signals and the voice data (DTMF tones). Requirement for Inband signalling is an uncompressed codec (G.711).

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).

Local echoes and reverberation effects are comepnsated.

(Voice Activity Detection)

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

The header is not set (recommended).

Use this option only tentatively for known problems with silent suppression.

Use this option only tentatively for known problems with silent suppression.

(recommended)

Comfort noise is offered during SDP.

Comfort noise is not offered during SDP. This option is only needed in case of compatibility problems with comfort noise.

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.

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

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

Alternative exchange line replacement tone.

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 call handling is defined by the codec negotiation with the 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.