FreePBX Announcement

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; This should will work for all IP7 devices.

Hardware #

This are the parts I was given for this project

• Intercom: Digital Acoustics IP7-SS20
• Speaker: Nippon Tc55 100w Power Horn
• 1 3.81mm centers 2 pin DIN connector
  • For connecting speaker to intercom
• USB-A to USB-B Connector
  • For connecting intercom to PC for Flashing

My intercom came with the power adapter.

Intercom Software Update #

My hardware's software was out of date, ~v5.3.3.4, and eSip doesn't seem to be an option until v6.3.4.23 on the SS20.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](https://www.digitalacoustics.com/talkmaster-focus-software-downlo ad/) 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](https://www.digitalacoustics.com/wp-content/uploads/software/S AFMA.exe))

The firmware files can be found ip7-firmware-download, it's packaged into an exe

Important: must be updated to v5.3.5.4 first.

The manual for SAFMA is located ip7-firmware-download, 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 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 6-7 factory resets to 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](https://www.digitalacoustics.com/talkmaster-focus-software-downlo ad/) 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 eSIP Standalone Configuration 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 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 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.

More work needs to be done 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.

Edit: markdownlint, textlint and meta TOML -> YAML