Help Profiles page This page lets you easily set the application for a given purpose. When a predefined setting is selected, the nonrelevant settings are set to defaults, so use this feature with caution. After applying the predefined settings, the SIP client can be additionaly customised for the shown properties only. All profiles have in common access to the complete set of settings from the following sections: - SIP Protocol Settings - Door Settings - Control Interfaces Settings - Notification Messsage Settings In addition, each profile provides extra customisation settings. The following profiles are available: SIP Phone This is the typical use of the SIP client. In this mode you can: Establish a full-duplex phone communication; Call up to 16 (with X8 attached) IDs; Select the audio input; The following additional options are enabled in this mode: - Close on timeout; - Close on Input 0-7 and X8 Input 0-7; - Input audio buffer level; - Input audio source; - Encoding; - Volume; - Mic and AD gains SIP Paging End Point In this mode the SIP client is used as an end point for getting SIP Paging messages. The typical use scenario in this case is to configure the SIP server to call a group of SIP clients which will autoreply and play locally the sent audio. While in idle mode, the SIP paging station may play background music. The following additional options are enabled in this mode: - Encoding; - Volume; - BGM IP Address and Port; - BGM volume and input audio buffer SIP Paging Gateway In this mode the SIP client is used to rebroadcast the incoming call to a specific multicast address, and port number. All Outbound Calls related settings are reset to the factory defaults. The following additional options are enabled in this mode: - Encoding; - Volume; - Audio Rebroadcast Address and Port. SIP Door Station The device is to be used as a door intercom station. Pushing the button causes the device to ring, and closes the call automatically after some time if not answered. In this mode the SIP client can be used in half duplex mode with AI Phone door panels. The phone pickup mode is preset to "auto answer" and cannot be changed. The following additional options are enabled in this mode: - Close on timeout; - Close on Input 0-7 and X8 Input 0-7; - Input audio buffer level; - Input audio source; - Encoding; - Volume; - Mic and AD gains; - Talk mode (HDX or FDX); - Output trigger level and trigger level timeout for the Voice Activity Detection (VAD); - AI Phone support (On/Off) SIP Monitoring Point In this mode the SIP client is used to call a predefined number if the input audio level exceeds certain level. As here only the Call/Close on level options are used, all IDs to be called on digital input are cleared (reset the factory defaults). The audio source input is preset to "Mic" and cannot be changed. The following additional options are enabled in this mode: - Call on level (Yes/No); - Call on level ID; - Level Threshold; - Close on Level (No/Time in seconds); - Close on timeout (No/Time in seconds);