posted Aug 9, 2018, 10:26 AM by EICSS Info
[
updated Aug 13, 2018, 6:23 AM
]
IRIS3000 board is shipped mounted with two rows of female connectors to plug-in Orange Pi Zero as drop down module. However, Orange Pi Zero only comes with one row of DIL male header. Before mounting Orange Pi Zero on IRIS3000, please make sure you have soldered a 13-pin SIL male header on available space on bottom side of the board. After mounting Orange Pi Zero on IRIS3000 and connecting to LAN, power it on using DC-pin power connector on IRIS3000. Caution: don't power the board using Orange Pi Zero micro USB connector. For a smoother trouble free operation, use a high quality 5V DC power supply rated at 2A. Once the board boots up, SSH to Orange Pi using provided credentials and proceed as follows to configure and run the firmware:
Hardware Specific Configurations Enabling FXO/FXS ports
You can enable/disable individual ports by setting corresponding parameters in cpe.conf:
# Enable/disable FXS
fxs-device enable
# Enable/disable FXO
fxo-device enable
Configuring USB Audio
IRIS3000 uses on-board USB voice codec for voice communication. First, make sure that the audio device has been successfully detected by Linux, using lsusb command:
The entry with PCM2900C is the entry you are looking for. If you don't see such an entry, the power supply is noisy. Use a good quality, preferably filtered, power supply and reboot the board. If USB device is there, run aplay -l command to find out card number of the device.
In the above example, the USB device is mapped to card 2, device 0. Therefore, set audio device in cpe.conf:
# Audio device
audio-device plughw:2,0 SIP Configuration
To register with Asterisk, modify SIP settings in cpe.conf to point to on-board Asterisk. Assuming you have created a SIP extension 200 with password 200, the setting would like:
# user password server port
sip 200 200 127.0.0.1 5160
FXS/FXO Ports Configurations
These are setting for call progress tones and CID. The settings are self explanatory. For FXS, you can keep default settings. For FXO, the call progress tones are for US. You may want to change these to match your regional settings. You may also want to change CID settings to match your land line configuration. Default CID settings are for the US:
# CID detection standard (dtmf, bell, etsi, none)
fxo-cid-standard bell
# CID detection alert (ring, dt, lr, none)
fxo-cid-alert ring
# CID detection wait time
# Value 0-30. Change it to tune to CID on your line
fxo-cid-wait 15
Please pay attention to last configuration item fxo-cid-wait. This is delay in 100 ms units which CID detector should use before starting decoding CID. You may have to tweak this value to match your service provider CID signaling.
The last item in FXO configuration is fxo-forward. This is for forwarding incoming call on FXO to DID of inbound route you will create in FreePBX:
fxo-forward 2000@127.0.0.1
Here 2000 is DID of inbound route.
Testing FXS Port
Once you are done with configurations, run the firmware in the folder where you have extracted files from firmware archive:
sudo ./cpe ./cpe.conf
If everything goes well, you should see initialization messages and finally a CLI prompt. Enter command ippconfig to check SIP registration status:
In the above example, the registration status is Pending. For a successful registration, it should be Ok. If registration is successful, you should be able to dial a feature code or any other extension registered to Asterisk on board.
Testing FXO Port
For FXO port, you need to integrate IRIS3000 firmware with FreePBX/Asterisk. Please follow the steps in the following section to configure FreePBX for both incoming and outgoing calls over FXO.
Integrating IRIS3000 firmware with FreePBX/Asterisk
To avoid dependency on complex kernel modules, IRIS3000 firmware is user
space application which integrates IRIS3000 with Asterisk as a custom trunk. Please follow instructions below to get a complete appliance up
and running. Please note that these
instructions are based on version 13 of both Asterisk and FreePBX.
FreePBX/Asterisk Configuration
First, we need to create a custom trunk. Go to Connectivity->Trunks and
create a custom trunk as shown below. Name it anything and keep all default settings in 'General' tab.
Next, go to 'custom Settings' tab and type in 'Custom Dial String' as shown below:
To use this trunk for outbound calls on FXO interface, go to Connectivity->Outbound Routes and create an outbound route:
Name it anything and keep default values in 'Route Settings' tab except setting 'Trunk Sequence for Matched Routes' to the custom trunk you created. Next, go to 'Dial Patterns' tab and provide a dial pattern of your choice. We chose to consider any number starting with 81 to be an outgoing number as shown below:
Note number 81 in prefix and '.' in match pattern. This means any number following 81 should be passed on to IRIS3000 for outbound dialing on FXO.
This completes your configurations for outbound calls on FXO. Next, we need to configure inbound calling. For that, go to Connectivity->Inbound Routes and create an inbound route:
Again, name it anything. Enter a DID Number of your choice. We chose to set it to 2000 and this is what we used to point fxo-forward to in cpe.conf as mentioned earlier. You also need to Set Destination to one of the destinations you want incoming on FXO to land to. This can be an extension, an IVR etc. We chose to ring an extension 102.
Almost done now except some configurations in Settings->Asterisk SIP Settings. Without these settings, Asterisk wouldn't let calls with a PSTN caller ID in. First, in 'General SIP Settings' tab, set 'Allow Anonymous Inbound SIP Calls' to 'Yes' and then in 'Chan SIP Settings' tab, set 'Allow SIP Guests' to 'Yes'. That's it. You should now be able to use both FXS and FXO ports of IRIS3000.
Troubleshooting
You may want to turn on debug messages in the CLI if things are not working as expected. This can be done by entering trace command in the CLI. Please report the trace messages when reaching out for support.
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posted May 19, 2018, 11:17 AM by EICSS Info
[
updated Aug 13, 2018, 6:25 AM
]
IRIS4000 board is shipped mounted with two rows of female
connectors to plug-in Orange Pi Zero as drop down module. However,
Orange Pi Zero only comes with one row of DIL male header. Before
mounting Orange Pi Zero on IRIS4000, please make sure you have soldered a
13-pin SIL male header on available space on bottom side of the board.
After mounting Orange Pi Zero on IRIS4000 and connecting to LAN, power
it on using DC-pin power connector on IRIS4000. Caution: don't power the
board using Orange Pi Zero micro USB connector. For a smoother trouble
free operation, use a high quality 5V DC power supply rated at 2A. Once
the board boots up, SSH to Orange Pi using provided credentials and
proceed as follows to configure and run the firmware:
Hardware Specific Configurations Enabling FXO ports
You can enable/disable individual ports by setting corresponding parameters in cpe.conf:
# Enable/disable FXO
fxo-device enable
# Enable/disable FXO2
fxo2-device disable
Configuring USB Audio
IRIS4000
uses on-board USB voice codec for voice communication. First, make sure
that the audio device has been successfully detected by Linux, using lsusb command:
The entry with PCM2900C is the entry you are looking for. If you
don't see such an entry, the power supply is noisy. Use a good quality,
preferably filtered, power supply and reboot the board. If USB device is there, run aplay -l command to find out card number of the device.
In the above example, the USB device is mapped to card 2, device 0. Therefore, set audio device in cpe.conf:
# Audio device
audio-device plughw:2,0 SIP Configuration
IRIS4000 integrates with FreePBX as PJSIP trunk. To be able to pass original caller ID to Asterisk, device authentication is disabled in the trunk as mentioned in subsequent section. Therefore, the firmware doesn't need to register with Asterisk as a SIP client. However, there has to be an inactive entry to satisfy the firmware. The default entry in cpe.conf looks like this and should not be changed:
# user password server port enable
sip 0 0 127.0.0.1 5060 0
The last parameter indicates that no registration message will ever be sent to Asterisk.
FXO Ports Configurations
These
are setting for call progress tones and CID for both FXO ports. The settings are self
explanatory and default settings are for the US. You may want to change these to match your
regional settings. You may also want to change CID settings to match
your land line configuration. Default CID settings are for the US are:
# CID detection standard (dtmf, bell, etsi, none)
fxo-cid-standard bell
# CID detection alert (ring, dt, lr, none)
fxo-cid-alert ring
# CID detection wait time
# Value 0-30. Change it to tune to CID on your line
fxo-cid-wait 15
Please pay attention to last configuration item fxo-cid-wait.
This is delay in 20 ms units which CID detector should use before
starting decoding CID. You may have to tweak this value to match your
service provider CID signaling.
Last items in FXO configuration are fxo-forward & fxo2-forward.
These are for forwarding incoming call on FXO ports to DID 2000. If you have configured a different DID value in FreePBX as described in subsequent section, change the value accordingly. The default configurations look like:
# FXO forward
fxo-forward 2000
# FXO2 forward
fxo2-forward 2000
Troubleshooting
You may want to turn on debug messages in the CLI if things are not working as expected. This can be done by entering trace command in the CLI. You may want to report the trace messages when reaching out for support.
Integrating IRIS4000 firmware with FreePBX/Asterisk
To avoid dependency on complex kernel modules, IRIS4000 firmware is user space application which integrates IRIS4000 with Asterisk as a PJSIP trunk. Please follow instructions below to get a complete appliance up and running assuming you have already installed Asterisk and FreePBX on the device and have latest IRIS4000 firmware. Please note that these instructions are based on version 13 of both Asterisk and FreePBX.
FreePBX/Asterisk Configuration
For a fully capable appliance which can pass through incoming CID as it, you need to create a pjsip_trunk. Go to Connectivity->Trunks and create a pjsip trunk as shown below: 
Go to pjsip Settings tab and enter data as shown below: Submit the changes and you will have a pjsip trunk named IRIS4000_PJSIP. Next, you need to create outbound route (Connectivity->Outbound Routes) as shown below: The most important item here is 'Trunk Sequence for Matched Route' which should be set to IRIS4000_PJSIP trunk created in previous step. Another important thing is Dial Patterns as show below: We have chosen to initiate all outbound calls over this trunk using prefix *81. The ',' in match pattern indicates that any number of digits are allowed and would be passed on to the trunk. Please feel free to chose any suffix that suites you. Next, you need to create and inbound route (Connectivity->Inbound Routes): We have named inbound route IRIS4000_INBOUND but feel free to name it anything. The most important item here is DID number which has been set to 2000. Again, this can be any number as long as the same is referred to in the firmware (discussed later). Another important item is 'Set Destination'. You can land incoming call to a number of destinations. Here, we opted an extension but it can be a ring group, IVR or any other option. Feel free to experiment. Once these three items are configured, Apply the settings using red button in upper right corner and you are done preparing FreePBX for iris4000. Firmware Configuration
The most important item in firmware configuration for integration with Asterisk is DID Number. If you chose to set to 2000 as shown above, you don't need to change anything in cpe.conf. If you did change it, just change fxo-forward and fxo2-forward configuration items to point to that DID. That's all. Run the firmware with configuration file as parameter and you should be able to successfully route both inbound and outbound calls from/to IRIS4000. |
posted Feb 2, 2018, 10:23 AM by EICSS Engineering
For customers who have got access to OPBX image, please follow the steps below to get a fully working FreePBX+Asterisk on Orange Pi Zero:
1- Flash the image on to an SD card. The image can fit in a 4 GB card but that wouldn't left enough space for data. An 8 or 16 GB card would be better. 2- Boot from the card and resize the partition to span all available space using resize2fs utility. 3- Reboot and run following command to fix a little bug which prohibits saving configuration changes in FreePBX: chown -R asterisk. /var/run/asterisk/
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posted Jan 5, 2018, 10:24 AM by EICSS Info
[
updated Feb 2, 2018, 10:25 AM by EICSS Engineering
]
IRIS4000 is our latest telephony hardware with two FXO ports. The configuration file i.e. cpe.conf looks pretty much identical to that for IRIS2000/IRIS3000 with some minor changes. We will be discussing these changes here for proper configuration of IRIS4000 firmware. Hardware specific configurationsSince 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 configurationsThis 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.
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posted Mar 17, 2017, 11:28 PM by EICSS Engineering
[
updated Jan 5, 2018, 9:53 AM by EICSS Info
]
Last time, we discussed integrating IRIS2000 with Orange Pi with a focus on Orange Pi One. In the mean time, Orange Pi Zero has become available. This board is an ideal choice as low cost headless computer as it is even smaller footprint and there are two good additions. One, is that it has USB signals available on a connector, making it easy to develop production grade hardware which rely on USB - no more USB cables. Second, it has WiFi making it a real value for money.
Orange Pi Zero is based on Allwinner H2 SoC compared to H3 SoC on Orange Pi One. Besides similarities between the two, there are differences - GPIO mapping is different event though 26-pin GPIO connector is RPi compatible. This means a different variant of Linux distribution (armbian) as well as WiringOP.
For Orange Pi Zero, relevant armbian version can be downloaded here:
https://www.armbian.com/orange-pi-zero/
Relevant WiringOP version can be downloaded here:
https://github.com/xpertsavenue/WiringOP-Zero
That's all. IRIS2000/IRIS3000/IRIS4000 should now run flawlessly with Orange Pi Zero.
Note: This article is an addendum to a previous article about integration with Orange Pi One located here.
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posted Nov 23, 2016, 9:52 AM by EICSS Engineering
[
updated Dec 20, 2017, 7:28 AM by EICSS Info
]
Introduction
Orange Pi is a low cost drop-on replacement of Raspberry Pi. As powerful as Raspberry Pi 2 and with a retail price tag of less than $10, it drives down product cost significantly. Keeping this in mind, we decided to get IRIS2000 working with Orange Pi. The model we choose was Orange Pi One which is by far lowest cost model:
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:
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 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!
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posted Sep 28, 2016, 11:35 AM by EICSS Engineering
[
updated Dec 20, 2017, 7:31 AM by EICSS Info
]
This CLI based demo allows initiating and receiving SIP calls. The demo binary is licensed to both individual and commercial customers free of cost. Please read through the license terms before you download and use the demo.
We intend to keep the demo updated as we add new features and fix bugs. Limited number of feature requests can be entertained provided they feel fit in general for all audience. Please send your feature request to sales@eicss.com.
Tested platforms
Raspbian Jessie (Raspberry Pi 1/2/3)
Prerequisites
- USB audio dongle
- ALSA run-time libraries
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.
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posted Aug 15, 2016, 11:28 AM by EICSS Engineering
[
updated Jan 5, 2018, 10:29 AM by EICSS Info
]
There are two aspects to integration with Asterisk:
- 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.
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posted Aug 13, 2016, 11:04 AM by EICSS Engineering
[
updated Dec 20, 2017, 7:34 AM by EICSS Info
]
You have received your board and now need to play around with it. Here are the items you need before you proceed:
- 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)
Here are the steps you need to follow:
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:
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.
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posted Aug 10, 2016, 10:01 AM by EICSS Engineering
[
updated Aug 15, 2016, 10:48 AM
]
IRIS series of telco hardware was originally designed for Raspberry Pi. IRIS1000 was designed for Raspberry Pi Model B Rev 2. Availability of I2S signals on a separate connector (P5) allowed use of I2S codec for audio. The board was eventually not directly compatible with later versions of Raspbery Pi and had to be redesigned. For full forward and backward compatibility, we decided to use USB codec. This gave birth to iris2000 which is compatible with all versions of Raspberry Pi.
Both versions of hardware can be used with other low cost computers. In fact, low cost products are probably more feasible with a sub USD 15 computer than RPi as this can substantially reduce product cost (RPi zero is another option but it has limited processing power to single core processor). IRIS1000 hardware interfaceAudio interfaceStereo I2S interface with following audio format: PCM, signed, 16-bit One channel of I2S is mapped to FXO while the other is mapped to FXS. GPIO interfaceThree GPIO pins are required for FXO while two GPIO pins are required for FXS interface. Thus a total of 5 GPIO pins are required for status/control of analog interfaces. A couple of more GPIOS may be used to indicate current status of the interface (used/unused etc.). IRIS2000 hardware interfaceAudio interfaceUSB audio interface with following audio format: PCM, signed, 16-bit One channel of I2S is mapped to FXO while the other is mapped to FXS. GPIO interfaceGPIO requirements are same as for IRIS1000. IRIS1000/2000 software interfaceAudio
In both cases, the audio interface can be accessed using Linux ALSA library. GPIOGPIO can be accessed using platform specific GPIO library. In Raspberry Pi Linux distros, WiringPi library can be used to access GPIO. Detailed information on exact GPIO mappings is available to hardware design licensees.
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