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Other Options

Allow inbound Calls

Location: SIP Trunks → SIP Trunk → Options tab → Call Options section

This option allows the SIP trunk to service incoming calls. If this option is deselected, any incoming calls will receive a “600 Busy Everywhere” error message. The call will be rejected by the PBX.

Allow outbound calls

Location: SIP Trunks → SIP Trunk → Options tab → Call Options section

This option allows the PBX to service outgoing calls through the SIP Trunk. If this option is deselected, 3CX will not allow any outgoing calls the be made through this SIP Trunk.

Disallow video calls

Location: SIP Trunks → SIP Trunk → Options tab → Call Options section

If the provider supports video calling and video codecs, this option, when selected will prevent video calls from being processed.

This is done by not sending any video codecs in the INVITE message to the provider, despite what might be in the initial request from the INVITE. Any video codecs in the initial INVITE from the endpoint will be stripped.

PBX Delivers Audio

Location: SIP Trunks → SIP Trunk → Options tab → Advanced section

This option forces the audio traffic to be redirected through the PBX, which acts as an audio proxy, instead of going from one endpoint to another endpoint directly. The PBX will force the audio to go through itself irrespective of which IP Address the SIP endpoints are requesting the audio traffic to go to.

This option is always recommended to be enabled.

Supports Re-Invite

Location: SIP Trunks → SIP Trunk → Options tab → Advanced section

The Re-Invite option will allow for a SIP endpoint to send an INVITE in an already established call, while keeping the same Call-ID.

The most common use for this is putting calls on hold and retrieving them from hold, where a re-invite is sent, changing the SDP parameters indicating that media attributes will change, all the while keeping the call active.

In most cases, this option for SIP Trunks is recommended to be disabled.

Supports Replaces

Location: SIP Trunks → SIP Trunk → Options tab → Advanced section

This option enables the support for the REPLACES header, which assists with the transferring of calls, for example, to and from a Shared Parking place or to perform an attended transfer.

This is used by the REFER method (RFC3515) and 3CX may use this when an external call is transferred to another external number.

In most cases, for SIP Trunks this option is recommended to be disabled.

SRTP

Location: SIP Trunks → SIP Trunk → Options tab → Advanced section

When SRTP is enabled, the audio traffic between the PBX and the SIP Trunk will be encrypted as per RFC4568. When SRTP is enabled on the trunk, during the negotiation of the call, 2 Media Streams will be sent in the SDP, one for encrypted traffic and another for the unencrypted.

This allows the SIP Trunk provider to use whichever of the 2 streams, either the SRTP or non-SRTP.

On incoming calls, when this option is enabled, 3CX will prefer to use an SRTP Media Stream if it is available in the incoming INVITE. If no SRTP Media Stream is present, it will use a non-SRTP one if that is available.

Number of SIM Calls

Location: SIP Trunks → SIP Trunk → General tab → Trunk Details section

This option will allow you to restrict the number of calls being processed through a SIP Trunk. This number is usually the number of channels the SIP Trunk provider allows for the specific account.

The number of simultaneous calls per SIP Trunk is the sum of inbound and outbound calls at any given point in time that are using the specific SIP Trunk.

When the maximum number of simultaneous calls per Trunk has been reached, 3CX will not allow any more outgoing calls through this SIP Trunk.

At the same time, If an incoming call is received, 3CX will reject the the call with a “600 Busy Everywhere” error message, similar to the message received when the incoming calls are restricted.

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