Hardware specific configurations

Since IRIS4000 has two FXO ports instead of one FXS and one FXO on IRIS2000/IRIS3000, the section has configuration parameters to enable/disable the two ports. These look like this:

# Enable/disable FXO
fxo-device enable

# Enable/disable FXO2
fxo2-device enable

These configuration parameters allow you to enable/disable any or both FXO ports.

FXO port configurations

This section has call progress tones configurations as well caller ID related configurations which apply to both FXO ports. The last two configuration parameters tell the firmware where to forward the inbound calls on FXO ports:

# FXO forward - extension where inbound on FXO
# should be forwarded to
fxo-forward 101@192.168.100.101

# FXO2 forward - extension where inbound on FXO2
# should be forwarded to
fxo2-forward 102@192.168.100.101

In this case, inbound on FXO is being forwarded to an extension 101 on local network while inbound on FXO2 is being forwarded to extension 102. Remember, however, that if you are using Asterisk or similar software and it's running on the same hardware the firmware is running on, the IP addresses above should be replaced with local loopback address 127.0.0.1.

Note: IRIS4000 firmware has a built-in call router. It keeps track of FXO port usage and routes outbound call to available FXO port. If no FXO port is available, the call is rejected.

Introduction

The best thing about Orange Pi One is that its GPIO connector is compatible with Raspberry Pi thus making the transition painless. Once you have the board and other accessories (power supply, USB cable, SD card etc.), follow the steps below to get IRIS2000 working with this low cost computer.

Step 1: Mount the iris2000 on Orange Pi

IRIS2000 mates with first 26 pins of Orange Pi One GPIO while for audio interface, a M-M USB type A cable is used to connect IRIS2000 USB port to one and only USB port on Orange Pi One.

Step 2: Setting up Linux environment

Get a good Linux distro. We tried various but found debian jessie from armbian to be the best. You can download it from this page:

https://www.armbian.com/orange-pi-one/

Extract the image, burn it to an 8 GB SD card and insert it into micro SD card slot in Orange Pi board. Power up the board. If everything goes well, you should be able to SSH to the board with following default credentials:

Login: root

Password: 1234

You will be asked to change password on first boot as well as create a user account.

Note: You should be able to use keyboard and mouse if you have connected HDMI display. However, we didn't see any activity on LCD and assumed either cable was faulty or the interface on the board itself was to blame.

Step 3: Installing required libraries

As for Raspberry Pi, you need to install two libraries required by iris2000 firmware:

• libasound2-dev
• wiringPi

libasound2-dev is standard ALSA library and can be installed with apt-get. For Orange Pi, a modified version of wiringPi called WiringOP is available here:

https://github.com/zhaolei/WiringOP

The installation process is quite straight forward, just clone the repository and build:

git clone https://github.com/zhaolei/WiringOP.git -b h3 && cd WiringOP

chmod +x ./build

sudo ./build

That's it. The library will be installed and you can run all wiringPi tests.

From here on, you can verify ALSA device number of USB codec and update cpe.conf accordingly.

Conclusion

Getting IRIS2000 working with Orange Pi One was difficult but fun. We had to try a number of Linux distros before we settled down on armbian. The efforts paid up and we have a even lower cost supported platform.

That's all for now. Good luck with your next Pi (Raspberry or Orange) project. Feel free to ask questions!

Running the demo

Please follow steps below to run the demo in popular ARM based platforms.

Step 1: Extract the archive

Extract the archive to a temporary folder:

tar xvf armphone.tar.gz

Following two files will be extracted:

armphone - executable

armphone.conf - Configuration file

Configure the phone

Edit the configuration file and change following configurations to your local settings:

audio-device - Provide USB audio device card and subdevice

net-device - Network interface to bind to

sip - SIP account info

bind - Local SIP port to bind to

avpf - Enable if your service requires AVPF instead of AVP in SDP

Run the executable

Run armphone with configuration file as parameter:

sudo ./armphone ./armphone.conf

If executable runs without any errors, it should produce an output as shown below:

You can find out current registration status using command ippconfig:

Or type help to list available commands.

That's all for now. Most of the commands are self explanatory. The phone also offers UDP based API which can be used to hook it up to a custom UI. Details of the API will be provided in another session.

• PSTN to VoIP call
• VoIP to PSTN call

Both the scenarios require configuration either in IRIS2000 configuration, Asterisk configuration or both. These are discussed below.

Configuration for PSTN to VoIP call

For an inbound analog call on FXO, fxo-forward command in cpe.conf needs to be configured to an extension where analog call can be routed. For example, if 101 is the extension where analog call should land, fxo-forward would look like

fxo-forward 101@127.0.0.1

Please note assumption that Asterisk is running on same RPi. If Asterisk is running on a different machine on your network, replace loopback address with IP address of your Asterisk machine.

Configuration for VoIP to PSTN call

To let Asterisk route dialed numbers matching preset dial rules, we need to add custom trunk in Asterisk and point it to IRIS2000. To do this in FreePBX, select 'Trunks' from 'Connectivity':

Select 'Add Custom Trunk' from list of trunk types:

The most important part is the dial string which should point to IRIS2000 on localhost. In case FreePBX is running on a different machine, change 127.0.0.1 to IP address of that machine.

Please note that in this example, IRIS2000 firmware was configured to listen on a different port (5091) than standard SIP port 5060 to avoid conflict with FreePBX which itself is configured for port 5060. You can use a port of your choice for IRIS2000 as long as there is no conflict.

That's all for integration with Asterisk. For more information about configuring dialing rules and trunks, please refer to FreePBX documentation.

• USB power supply rated at 2A
• USB Type A M-M cable
• SD card with Linux distro (Wheeze/Jessie/Ubuntu) installed
• A Raspberry Pi computer (Pi 1/2/3/Zero)
  

Step 1: Mounting the board on Raspberry Pi

Being backward compatible, the board mates with first 26 pins of GPIO while for audio interface, a M-M USB type A cable is used to connect IRIS2000 USB port to one of USB ports on Raspberry Pi as shown below.

Step 2: Setting up Linux environment

Power on the board and let it boot to completion. I'm assuming the Pi is connected to your local network using Ethernet port or WiFi. Once you discover IP address of the device, you can connect to it using SSH. Alternatively, you can connect a display and keyboard to the Pi. Following libraries are required to run demo firmware:

• libasound2-dev
• wiringPi

Once these libraries are installed, you can run demo firmware but before you do that, check mapping of USB audio device using aplay -l:

This will list all ALSA compatible devices including IRIS2000 USB audio device. If you don't see the device in the list, verify that USB codec has indeed been enumerated by lsusb command in bash as shown below.

The entry with PCM2900C is the entry you are looking for. If you don't see such an entry, make sure USB cable is connected correctly, is not faulty and that you are using a high quality USB power supply as USB devices are not detected if power supply is noisy.

Step 3: Configuring firmware

We need to configure the firmware before running the firmware. Simply edit the configuration file named cpe.conf in any editor and follow the comments to configure the firmware to your requirements.

Step 4: Running firmware

We are now ready to run the firmware. The firmware file name is cpe and should be run with sudo. The firmware assumes that configuration file named cpe.conf resides in the same folder the firmware is being executed from.

All is well if the firmware runs successfully all the way to a command shell. Depending upon your configurations (standalone analog-to-VoIP call router or integrated with Asterisk), you can go ahead and test out analog-VoIP call bridging.

That's all for now. A separate post discusses IRIS20000 integration with Asterisk.