[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.
There’s something so nostalgic about the rotary phone that makes it a fun thing to hack and modernize. [Voidon] put his skills to the test and converted one to VoIP using a Raspberry Pi. He used the RasPi’s GPIO pins to read pulses from the rotary dial – a functional dial is always a welcome feature in rotary phone hacks. An old USB sound card was perfect for the microphone and handset audio.
As with any build, there were unexpected size issues that needed to be worked around. While the RasPi fit inside the case well, there was no room for the USB power jack or an ethernet cable, let alone a USB power bank for portability. The power bank idea was scrapped. [voidon] soldered the power cord to the RasPi before the polyfuse to preserve the surge protection, used a mini-USB wifi dongle, and soldered a new USB connector to the sound card. [Voidon] also couldn’t get the phone’s original ringer to work, so he used the Raspberry Pi’s internal sound card to play ringtones.
The VoIP (SIP) was managed by some Python scripting, available at GitHub. [voidon] has some experience in using Asterisk at his day job, so it will be interesting to see if he incorporates it in the future.
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.
[Daniel] picked up a cheap USB handset to use with his VoIP provider, and included in the box was a CD with all the software that would make this handset work with Windows. [Daniel] is running Linux on his main battlestation, rendering the included CD worthless. Using the handset under Linux would be a problem; although the speaker and mic worked, the buttons and screen did not. No problem, then: [Daniel] just played around with the command line until he figured it out.
The handset presented itself to the Linux box as a soundcard and HID device. The soundcard was obviously the speaker and mic, leaving the buttons and display as the HID device. [Daniel] checked this out by running a hexdump on the HID device and pressed a few buttons. His suspicions were confirmed, and he could easily read the button with a little bit of Python.
With the speaker, mic, and buttons on the handset figured out, [Daniel] turned his attention to the one bit of electronics on the phone he hadn’t yet conquered: the display. After firing some random data at the phone, the display blinked and showed a messy block of pixels, confirming the display was controlled through the HID driver. Loading up usbsnoop to see what the original software does to update the screed showed [Daniel] the data format the display accepts, allowing him to control everything in this VoIP phone.
[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?