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This will be a general outline of how to setup a Digital Acoustics Intercom and Paging device as an stand-alone Sip extension. This may not work in all FreePBX configurations, but this is how I got to work in it’s most basic form; I believe this will work for all IP7 devices.

Hardware

This are the parts I was given for this project

My intercom came with the power adapter.

Intercom Software Update

My hardware’s software was very out of date, ~v5.3.3.4, and eSip doesn’t seem to be an option until v6.3.4.23 on the SS20.

Most of Digital Acoustics’ documentation tries to push you into using the TalkMaster Suite of tools. I personally had no luck updating using Firmware Management Console, which is supposed to update the firmware over a network connection. Instead I downloaded the Stand Alone Firmware installer (SAFMA) found in section 7 of the troubleshooting page ( Direct Download )

The firmware files can be found here , it’s packaged into an exe and I advise installing it in Documents

Important: must be updated to v5.3.5.4 first.

The manual for SAFMA is located here , generally it’s self explanatory:

  1. Open software
  2. Select Firmware
  3. Plugin Device using USB-A to USB-B Connector
  4. Select COM port (may auto do it)
  5. Reset device with Reset button
  6. Remove and Replace USB-B Connector from device
  7. Click Start

I found step 6 wouldn’t re-register the device as being reset and would prompt to restart again, which would fail. Doing a factory reset and trying again fixed this issue, but you would have to restart this process.

Factory Reset:

  1. Unplug device from power
  2. Hold Volume up and down
  3. Reattach to power
  4. Wait until beep (it will be at full volume)

This took me multiple factory resets to successfully update it. I’m not sure what the problem was.

FreePBX

The main point of doing the extension setup next is that the extension user password is required during the intercom configuration. Again: this worked in my particular case, this was the first time I used this particular software and results may vary.

Extension can be found in the Application section in the header. Once there click New Extension and Chan_Sip, Here are the settings I used:

  • General
    • Name: Intercom
    • Extension #
  • Voicemail:
    • No change, all disabled
  • Advanced
    • can reinvite: update
    • host: {Static IP}
    • Default user: {extension #}
    • NAT Mode: No

Now Save, but don’t apply config:

  1. Go back to General
  2. Scroll to User Section
  3. Click Linked to user {ext#} (page link)
  4. Change user password and take note of it

Now Apply Config

Intercom Setup

Instead of using the TalkMaster Suite Admin Console, I used the eSIP Configuration Utility instead. You might be able to do it through TalkMaster, but these instruction will be in the context of eSip Utility.

Note: If you installed TalkMaster and you want to configure using eSip, the TalkMaster service must be stopped as they both use port 3000. This is how you can do that:

  1. Open CMD (as Admin)
  2. Run: netstat -ano | findstr :3000 (pid will be last column)
  3. Run: talkkill /PID {enter_pid} /F

eSip Configuration

I was unable to find the eSIP Configuration manual, but here is the Admin Console’s

The first step is setting the Static IP. I presume this can be done through the Utility, but I prefer reservations. I took note of the device’s IP in the discovery, went to my DHCP server, found that IP, made a reservation with that MAC and finally reset the device.

Using the Utility, there are the settings I put in:

  • General
    • Standalone Sip
  • Sip
    • peer-to-peer
    • SIP + RTP (checked)
    • extension + auth is extension #
    • pass is extension user password
  • Advanced
    • Activate on PPT

Note: The manuals show using SIP Extension Only, but I was unable to get this to work as the device would always appear as unreachable. The only way I found success was using peer-to-peer

FreePBX Validating

Here are where the Sip logs can be found:

  1. Reports in the header
  2. Astrisk Info in the dropdown
  3. Chan_sip Info on the list to the right

Scrolling down to Chan_Sip Peers you should see the new extension and the furthest column on the right should say OK ({x} ms). Generally looking at this page is faster than dialing the extension on the phone.

Fin

When calling the extension the IP7-SS20 automatically picks up and plays the caller’s voice through the speakers.

I believe more work needs to go into this for it to be set up properly; Possibly a passcode or blocking specific phones from calling it, but neither are concerns in this setup right now.

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