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Difference between revisions of "New fields (AlphaPro 10.27)"

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* '''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.
 
* '''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.
 
* '''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 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).
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* '''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).
  
 
==== Timers ====
 
==== Timers ====

Revision as of 10:38, 28 August 2007

A number of new fields are added in the VoIP tab under Exchange & System -> System:

AlphaPro VoIP window

Flags

  • 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).

Timers

  • Max VoIP round trip delay allowed for open duplex during handset conversation in AlphaNet:
  • SIP digit collection timeout:
  • SIP Dial Delay:

Groupcalls, Audio programs to IP stations

  • Base Address:
  • Audio Program feed in master:
  • Relay address and Mask 0 - 4:



.ex_profile.glob_const.priv_ring_mail_pri .ex_profile.glob_const.bc_event_node