New fields (AlphaPro 10.27)
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Exchange & System -> System -> VoIP
- Use VoIP audio for Multi Module: Disable this flag when using AGA or AE1 for audio distribution in MultiModule, and not VoIP. Note that it is also possible to use a mix of VoIP audio and conventional AGA/AE1 links. Then this flag must be enabled.
- Optimized voice duplex control when conversation with SIP/trunk/stations: When calling SIP there can be problems with the standard duplex algorithm due to DSP echo cancelling in the SIP station. A new duplex algorithm is available for duplex towards SIP stations that is speech controlled only from volume of the microphone signal from the SIP link. When this flag is set, the initial voice direction is forced to be from the intercom towards the telephone. When the phone operator starts to speak, the voice direction will switch towards the intercom station, regardless of the level of the audio signal from the intercom station. As soon as the phone operator stops speaking, the voice direction will switch back to the initial direction.
- Incoming calls from SIP in private ringing mode: When enabled incoming calls from SIP trunk line or SIP phones will be forced in private ringing mode, regardless of the private/open switch of the intercom station (this flag is moved from Exchange & System -> System -> Calls and Options).
- Max VoIP round trip delay allowed for open duplex during handset conversation in AlphaNet:
- AlphaNet calls: The roundtrip delay on a VoIP AlphaNet conversation is measured by the AlphaCom. If the delay is more than the value specified in this parameter, a handset to handset conversation will be forced in voice switched duplex. This is to prevent echo in the handset due to crosstalk in AlphaCom handset (and no echo cancelling).
- SIP calls: When talking with handset conversation between Intergard station and SIP station the SIP station will get echo in the handset due to crosstalk in AlphaCom handset (and no echo cancelling). Due to this handset to handset communication will be forced in duplex. (Default delay setting for SIP = 30ms. Full duplex can be obtained if this parameter is adjusted to 40 ms or more)
- SIP digit collection timeout: Specifies how long the AlphaCom should wait for more digits before setting up the connection to a SIP device. The timeout is used by feature 81 and feature 83 when the field "Collect N more digits (SIP)" is used.
- SIP Dial Delay: After having made a connection to a SIP gateway, further dialing is delayed by the time specified by this timer.
Groupcalls, Audio programs to IP stations
Base Address and Relay Address are related to groupcalls, audio program and program conference to IP stations.
The group audio functions are based on multicasting the audio. In addition the software supports up to 5 ranges of IP addresses, which defines 5 subnets, which can get a copy of the audio as a unicast. One of the stations in each subnet get a unicast copy of the audio and messages, and will forward to local stations on multicast.
- Base Address: First multicast address used by this node. Default is empty. If empty 239.192.nnn.0 is used, where nnn is node number. Usually this default can be used. Only need to be changed if conflicts with other multicasting applications on the same net. Alphacom may use up to 31 adresses following the base-address. These adresses are used for group call, program conference and audio program to IP stations.
- Relay Address and Mask 0 - 4: List of five (0-4) address-ranges for IP stations that can receive a unicast relay (copy) of a multicast.
- Exchange reset needed after change.
- Note: IP station software ver. 01.06 is required for groupcalls, audio program and program conference!
- Audio Program feed in master: When this flag is enabled Audio Programs need only to be connected to the master in a MultiModule. When the flag is disabled Audio Program must be connected in parallel to all modules, as before. Exchange reset is needed after changing the flag.
- The "Audio program feed in master only" will work for all types of module interconnection (AGA / AE1 / VoIP). The MultiModule audio links are a pool of ressources available on demand for conversations, group call, conference or audio program. I.e audio links are not reseved for audio programs, but will be assigned whenever needed.
- When a station in a slave module selects an audio program (say 801), an audio link is set up from slave to master. As long as one or more stations in the slave is listening to this program, the audio link will stay active. If a second program (say 802) is selected in the slave, another audio link is activated. When nobody listen to the program any longer, the audio link will be released.
- The frequency range of the audio program in slaves will depend on the interface boards used:
- AGA board: 18,5 kHz
- AE1 board: 7 kHz
- VoIP: 7 kHz
- This is the bandwith between the AlphaCom modules. The bandwith in the station also depends on the bandwith of the station itself.
- Note that when using AGA via multiplexers, the bandwith depends on the audio coding used by the multiplexer. Often telephone bandwith (3,4 kHz) is used.
Exchange & System -> System -> Logs and Errors
- Broadcast event messages to remote node: ACDP broadcast messages will be forwarded to the specified node. Useful for Call Handler applications like AlphaVision used in AlphaNet, which relies on status information received from each individual AlphaCom exchange. Requires AMC sw version 10.21 or newer.
Exchange & System -> System -> Calls and Options
- Private ring mail priority: When receiving Mail or Call Request with this priority or higher, the receiving station will be alerted by a private ringing tone. Default priority level is 150. Also ATLB subscribers (telephones) can receive this type of Call Requests, as the telephone will start to ring when a call is received.