IP Audio Client Firmware V2.10.0
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This image can be installed ONLY on the following Barix devices running an IP Audio Client FW version greater or equal to V2.7.0:
Exstreamer M400
IP Former TPA400
Exstreamer MPA400
Annuncicom MPI400 (new entry from V2.10!)
Please don’t install this firmware on any of these products: SIP Opus Codec, Audiopoint, RetailPlayer, Soundscape, Store & Play
Change Log
New Features
ANNUNCICOM MPI400 SUPPORT:
IPAC from V2.10 is the official and only supported firmware for Annuncicom MPI400. Developments on this release are mostly focused on adding features supported mainly on this product
Digital Input Operation modes: Configurable individually for each dry contact input available
SIP: initiate calls, accept incoming calls, hangup calls when the input is closed.
SIP “Press-Hold-Release”: same as SIP with the difference that any state change of the input triggers the action to perform.
SIP Refer Blind Transfer: Transfer a call to the configured extension
Input operational modes and configuration of the extension to be dialed are assigned in the new IO Page on the web interface (only available on hardware with input and output capabilities)
Relay Output activation: Configurable individually for each dry contact input available
Relay activation on SIP RINGING state (de-activation when ringing is over)
Relay activation on SIP ACTIVE call state (de-activation when call is hangup)
Relay activation on SIP DTMF sequence (de-activation upon reception of DTMF de-activation sequence, configured separately)
Relay activation from REST API call (if enabled in the configuration)
Relay activation modes are configurable in the Relay section of the IO page (only available on hardware with input and output capabilities)
Audio Input management:
Added support for gain adjustment of an audio input interface in the settings page of the web interface:
Line In: to get the signal from the line level input
Mic: to get the signal from the standard mic input
Phantom Mic: to get the signal from the phantom powered mic input
Gain: to adjust the gain of the selected input
SIP Source
Added SIP ringing tone to be played when auto answer is off
Added SIP “Half Duplex” mode with “level” and “timeout” adjustments to enable fine tuning of “talk” and “listen” status
Added “Capture Buffer Latency” to adjust the input buffer of the SIP Client
Added “Audio Input Level” to fine tune the input level setting of the SIP client
Added “Auto-Answer” enable / disable option and “Auto-Answer timeout” to pick up calls automatically after the specified interval
REST API Extension:
Extended REST API with new calls to:
GET Input Status
GET Relay Status
SET (PUT) Relay Status (API control must be enabled on the intended relay)
Fixes / Improvements
Known Limitations / Issues
Configuration Tool: SIP Server settings related to username must be entered manually in the tool for each device despite they are copied from one device to all others when pushing the configuration or copying and inserting.
Configuration Tool: the green cell indicating the playing source is not updated in real time. It requires the user to click the READ ALL button.
RTP Regular mode: frequency drifts (playback speed adjustments) when playing formats with lower sample rates (e.g. G711 @ 8kHz)