My personal testing was performed on a FreePBX 13 system running Asterisk 13.
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These notes presume the reader is familiar with programming the FreePBX Asterisk server, including how to add a trunk and make it functional. The second, more involved, approach will work with an Asterisk server with a static IP address on the Internet interfacing with an OBI sitting behind a firewall that has a dynamic IP.
I will address two scenarios: The first, and simpler, implementation presumes that the Asterisk server and OBi device are on the same local LAN. Every configuration option described herein is based on publicly available published documentation. What follows is a description of how to configure Asterisk and the OBi device to accomplish this task. However, many of us already use these OBi units as ATAs in which case the additional functionality can often be added without any additional hardware costs. The OBi200 can be had for around $50 and as low as $35 on sale. In essence, the OBi200 or 202 ATA look like an ITSP to the Asterisk server and make it possible for Asterisk to continue to support up to three GV lines per OBi. And (2) incoming GV calls would be automatically translated into SIP and delivered to Asterisk. I was able to interface an OBi202 to an Asterisk 13 server such that (1) outgoing calls from Asterisk would flow into the 202 using the SIP protocol, where they would be bridged to a corresponding GV line to complete the call. After some experimentation I came to the conclusion that, yes, they can. Could they be pressed into service to provide a SIP-to-GV bridge? Think of this as a small localized version of Bill Simon's Google Voice Gateway. Polycom (neé OBiHAI) makes the only non-Google devices which currently have access to the new GV network with Google's blessing, and this got me thinking: The OBi 200 and 202 devices are themselves a feature-rich ATA - some might even say a mini-PBX in themselves - capable of handling four trunks and/or endpoint devices. In mid-2018, Google shut down its XMPP servers - the protocol the Asterisk PBX used to access GV.
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While it's certainly not a perfect fit, it can be made to work reasonably well and free offsets a multitude of sins. Many VoIP hobbyists and experimenters have come to rely on Google Voice for their trunking needs. Finally, feel free to critique these notes if you see areas where changes are needed or they can be improved. Let me know if you are successful (or not), along with any alterations you had to make, and I'll include your results here for the benefit of others in the community. I believe they should continue to work fine, perhaps with some minor tweaks. Feel free to make use of them within your own VoIP system, with the caveat that they have not been tested personally on systems beyond Asterisk 13.
However, I have unfortunately not kept up with the evolution of the Asterisk server, and as such these notes may well have become dated.
In addition, I have exchanged emails with perhaps a dozen other individuals who have successfully implemented their own bridges. I have successfully programmed many OBi devices to perform the SIP/GVsip bridge function described in these notes. Turning your OBi200 or OBi202 into a SIP-to-Google-Voice Bridge Turning your OBi200 and OBi202 into a SIP-to-Google-Voice Bridge Robert Stampfli - Published June, 2018 last updated Saturday, 08 December 2018
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