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SIP phone as station

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Revision as of 14:10, 30 May 2012 by Asle (talk) (→‎SIP station Configuration)
AlphaCom XE as SIP Server

This article describes how SIP users can be used with the AlphaCom XE servers. SIP users are related to "physical" numbers in the same manner as STENTOFON IP stations.

Examples of SIP users are:

  • VoIP telephones
  • PC clients (Softphones)
  • WiFi phones
  • IP Dect systems
  • Analog Telephone Adapter (ATA)

The AlphaCom XE supports up to 500 SIP users.

A "SIP station" license is required for each SIP phone. The license is available in stages of 1, 6, 12 and 36 SIP users.


Feature support

In principle SIP users have the same features and settings available as Stentofon stations. However, because of the limitations of the SIP phones and protocol, some features are not supported.

Feature list

Feature SIP Users STENTOFON Stations
Point to point calls V V
Caller ID V V
Duplex Conference V V
Door opening (6) V V
Send Call Request V V
Member of Ringing Group V V
Line Down reporting V V
Receive Call Request (Call Queuing) X V
Simplex Conference X V
Audio Program X V
Receive Group Calls X V
Receive Voice Alarm Messages X V
Receive Mail Messages (Station errors etc.) X V
Status info in display (Absence, Transfer etc.) X V
Volume setting X V
Hotline call X V
Line Monitoring via Tone test X V

About Line Down reporting

Line down reporting on SIP phones is based on the "Registration expire" time out. This timeout is default several hours on most SIP phones. If faster fault reporting is needed the timeout must be adjusted.

The timeout can also be adjusted as a general parameter in AlphaCom NVRAM: 

ex_profile.timeouts.sip_max_expire

0 = Use time out as defined in the SIP phones <br\> 0< = Time out in seconds

About Hotline Call

Hotline call is only available from SIP stations with best level of integration with option for "off hook auto dialling". <br\> When this is configured the SIP phone will get the AlphaCom dial tone when lifting the handset and the hotline timer is started.

Software requirements

  • AMC 10.56 or higher
  • AlphaPro 10.56 or higher

In AMC software prior to 10.56, SIP users had to register via a virtual SIP Registrar node. This was a more complicated solution, with less functionality.

AlphaCom Configuration

AlphaWeb Configuration

  • Licenses: Each SIP user requires a "SIP station" license. In AlphaWeb, go to System Configuration -> Licensing, and Insert the license key containing the SIP Station license.
  • Filter settings: The ports used for SIP protocol (5060) and the VoIP Audio must be enabled for the ethernet port used by the SIP users. In AlphaWeb, go to System Configuration -> Filters, and enable the UDP ports for SIP (5060) and for VoIP Audio (61000:61150). By default these ports are enabled on ethernet port 1.

AlphaPro Configuration

SIP phones are configured in the Users & Stations window in AlphaPro
  • From the Users & Stations window in AlphaPro, select a free user from the listbox, and enable the SIP station flag
  • Configure Directory Number and Display Text
  • Select a supported Codec for the SIP station to use


SIP station Configuration

SIP phones have various local set-up and configuration options. The level of integration will depend on the configuration available on the current phone model.

Best Integration Level: SIP phones supporting "Automatic dialing when Off-Hook" and SIP INFO digit signaling.

Automatic dialing when Off-Hook When lifting handset the SIP phone will do an automatic setup of a call. AlphaCom will give dial tone and interpret the digits dialed "live" giving immediate response, ASVP messages and event trigging of station in use etc. This will give same functionality as dialing digits from an ATLB-telephone.

SIP-phones with Auto-answer option for incoming call can also support the standard Intercom functionality "OPEN" mode when receiving call. <br\>

Medium Integration Level <br\> 

SIP phones with SIP INFO signaling but without automatic dialing when "off hook"

When starting a feature (make a call) digits must be collected in the SIP phone before sent to the AlphaCom for interpretation. This will not give any feedback to the user before the collected digits is sent to the AlphaCom in the “INVITE” message. (Usually trigged by pressing the handset button, timeout or a dedicated “send” key.). <br\> Further dialing after activation of a feature like Inquiry or programming of wake up will be supported by SIP INFO signaling. Event-handler events can not be activated before the SIP client sends the information. (Station in use etc. will not be trigged when the user starts to dial locally) <br\> Basic Integration Level <br\> 

SIP phones without SIP INFO signaling.

Activation of features will be supported in the same manner as Medium Integration level, but the SIPD only support digits during conversation sent as SIP INFO signalling. Stations without this type of signalling will not be able to do any feature activation during conversation or use features requiring extra parameters. In conversation functions like inquiry, transfer and search will not work. Features requiring extra parameters like WakeUp and Follow Me will not work.

Configure the SIP phone with the following parameters

  • Basic level must be programmed for all SIP stations.
  • Medium and Best level can only be programmed if supported in the SIP station.

Basic Integration

The IP address of AlphaCom
Directory number matching directory number as configured in AlphaPro
Codec matching the codec as configured in AlphaPro

Medium Integration

Digit signalling with SIP INFO signalling. 
DTMF signalling in Audio band must be turned off if SIP trunks are to be used.

Best Intergration

Configure the "auto off hook dialling" sequence with the text: HOOK <br\>
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