Even though the age for first carrying a smartphone seems to be decreasing, there’s a practical lower minimum age at which a kid can reliably use one to make a call. So how do you make sure your tot can reach out and touch mommy or daddy? This toddler-friendly Raspberry Pi hotline is a good start.
With a long trip to Hawaii pending and a toddler staying behind, [kuhnto] wanted a way to make communication as simple as possible. In the days of pervasive landlines, that would have been as simple as a feature phone with a couple of numbers on speed dial buttons. With nothing but cell phones to rely on, [kuhnto] turned to a Raspberry Pi running PBX software and a command line SIP client for making calls over a Google Voice line. The user interface is as simple as can be – a handset and two lighted buttons on a wall-mounted box. All Junior needs to do is pick up the handset and push green to talk to Daddy, blue for Mommy. Something similar might even be useful for elder care.
Kudos to [kuhnto] for thinking through the interface issues to come up with a successful build. We’ve seen other UIs simplified for kids before, such as this button-free jukebox or this special-needs media player.
[Alessandro] is an unlucky VoIP PBX administrator that frequently has to deal with very, very dumb network policies. Often times, he’ll have to change something on his setup which requires him to go out to his client’s location, or ask a client to use Teamviewer so the appropriate change can be made from behind a firewall.
This isn’t the solution to the problem. It will, however, fix the problem. To get around these firewalls, [Alessandro] is using the voice channels he already has access to for changing configurations on his VoIP boxes.
The implementation of this uses the AX.25 amateur radio modules that can be found in just about every Linux distro. This, and an Alsa loopback device, allows [Alessandro] to access a terminal over a voice-only network. Is it a hackey kludge? Yep. Is it just a little bit dumb? So are the network policies that don’t allow [Alessandro] to do his job.
This build isn’t too dissimilar than a bunch of modems from the old BBS days, albeit with vastly more powerful software. [Alessandro] says you’re only going to get about 38400bps out of this setup, but it beats begging for help for remote access.
The SIP protocol is commonly used for IP telephone communications. Unfortunately it’s notorious for having issues with NAT traversal. Even some major vendors can’t seem to get it right. [Stephen] had this problem with his Cisco WRVS4400N router. After a bit of troubleshooting, he was able to come up with a workaround that others may find useful.
The router had built in SIP ALG functionality, but it just didn’t work. [Stephen] was trying to route SIP traffic from a phone to an Asterisk PBX system behind the router. The router just couldn’t properly handle these packets regardless of whether SIP ALG was enabled or disabled.
[Stephen] first tried to change the SIP port on the external VOIP phone from the default of 5060 to something else. Then he setup port forwarding on the router to the Asterisk box to forward the traffic to the Asterisk system on the original port. This sort of worked. The calls would go through but they would eventually drop after about 20 seconds.
The only thing that [Stephen] could get to work completely was to change the SIP port in Asterisk’s sip.conf file using the “bindport” directive. He changed it to some random unused high port number. Then he setup port forwarding on the router to forward incoming UDP packets on that port to the Asterisk system. This worked fine, but now all of the original phones behind the router stopped working because they were configured to use the default port of 5060.
Rather than re-configure all of the phones in the organization, [Stephen] made one change on the Asterisk system. He setup an iptables rule to forward all incoming traffic on UDP port 5060 to the new SIP port. Now all of the phones are working with minimal changes across the organization. It’s a lot of hassle to go through just because the router couldn’t handle SIP correctly, but it gets the job done.
Payphones used to be found on just about every street corner. They were a convenience, now replaced by the ubiquitous mobile phone. These machines were the stomping grounds for many early computer hackers, and as a result hold a place in hacker history. If you’ve ever wanted to re-live the good ol’ days, [hharte’s] project might be for you.
[hharte] has been working to make these old payphones useful again with some custom hardware and software. The project intends to be an interface between a payphone and an Asterisk PBX system. On the hardware side, the controller board is capable of switching various high voltage signals required for coin-line signaling. The controller uses a Teensy microcontroller to detect the hook status as well as to control the relays. The current firmware features are very basic, but functional.
[hharte] also wrote a custom AGI script for Asterisk. This script allows Asterisk to detect the 1700hz and 2200hz tones transmitted when coins are placed into the machine. The script is also in an early stage, but it will prompt for money and then place the call once 25 cents has been deposited. All of the schematics and code can be found on the project’s github page.
The Pogoplug Series 4 is a little network attached device that makes your external drives accessible remotely. Under the hood of this device is an ARM processor running at 800 MHz, which is supported by the Linux kernel. If you’re looking to build your own PBX on the cheap, [Ward] runs us through the process. Since the Pogoplug 4 is currently available for about $20, it’s a cheap way to play with telephony.
Step one is to convert the Pogoplug to Debian, which mostly requires following instructions carefully. After the Pogoplug is booting Debian, the Incredible PBX bundle can be installed. We’ve seen this bundle running on a Raspberry Pi in the past. Incredible PBX’s preconfigured setup based on Asterisk and FreePBX gives a ton of functionality out of the box.
With your $20 PBX running, there’s a lot that can be done. Google’s Voice service allows unlimited free calling to the USA and Canada. With Internet connectivity, you get email notifications for voicemails, and can query WolframAlpha by voice.
It looks like a consumer good, but this PBX server blade was built by [Benoit Frigon] over the last couple of years. It brings multiple telephone extensions to his home service.
The device runs Asterisk open source PBX software. Because it will be on all the time he wanted something that doesn’t draw a lot of power. The 500 Mhz system seen on the left has just a half a gig of ram. It’s enough to do the job and at 10 Watts it’s not going to break the bank when it comes to paying the electric bills. The board in the middle is used to interface the analog handsets with the land line. From the look of it he’s got it rigged for two extensions.
That’s all somewhat par for the course with PBX rigs, but the enclosure is where he really shines. [Benoit] used 22 gauge aluminum sheet to fabricate the enclosure which is designed to blend in with the rest of his home’s rack mount hardware. To provide control at the rack he added his own LCD and touch-sensitive button interface to the front of the case based on a PIC 18F2520. The system can also be accessed via the web thanks to a custom interface he coded.
[Ward Mundy] has found something great by combining a GXP-2200 phone with Raspberry Pi to create a private branch exchange. So the idea behind a PBX setup is kind of like a company intranet. All of the phones in the system are assigned an extension number and have access to the internal system functions like voice mail, and sharing phone lines to the outside world. We’ve talked about using an RPi as a PBX before, but the high-tech phone he’s using this time around pulls everything together remarkably well.
The GXP-2200 is available for under $200. It runs Android and has a full color touch screen pictured above. It is marketed as a multimedia phone and indeed it brings Skype and Google Voice to the party. But it also offers six SIP lines. The hardware even seems to be planned for this type of use as the phone offers a second Ethernet port to which the RPi board can be connected. In this example [Ward] simply screws the RPi to the phone’s plastic stand and connects the two using a six-inch cable. From there the PBX can be configured with the phone’s browser. How’s that for slick?