R.I.P. Simonics GVSIP Gateway Going Away

wardmundy

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Some sad news from Bill Simon this morning:

THIS IMPORTANT SERVICE CLOSURE ANNOUNCEMENT IS BEING SENT TO ALL USERS OF SIMON TELEPHONICS GOOGLE VOICE GATEWAY.

You may have read that in late April, Google announced that they will be discontinuing XMPP Interop with Google Voice. See our news posting at »simonics.com/2018/05/fut ··· gateway/. Since that time, we have worked on an updated Google Voice Gateway implementation that would allow us to connect to Google using their new SIP-based protocol. From late May to mid-June, we ran a beta test of this new software to find bugs and determined that it would be suitable for production.

On Sunday, June 24, we attempted to put the new software into production. A serious problem was uncovered at this point related to the large number of accounts we are registering to Google Voice. On Tuesday, the system was rolled back and we studied the problem. Unfortunately, what we found was a very large increase of computing resources needed in order to support the new platform at production scale. We attempted some optimizations, but the problems are fundamental to the VoIP engine, and the resource requirements to run the new platform are not sustainable.

With regrets, I am announcing the discontinuation of Google Voice Gateway at the time when Google fully discontinues their XMPP interop. This was slated to begin on June 18, so we expect that the remaining time is short--perhaps another week. Thank you to all of you who have supported this service over its 6+ year history.

Accounts should be working at this time using the Google Chat (XMPP) connections. If you are not receiving calls, please see the post by me on this forum thread: »support.simonics.com/sup ··· 8/page/3 .

Users looking to continue service with Google Voice using regular phones should consider the Obihai 2xx, 10xx, and 2xxx series phones, or use the new Google Voice app for Android (wi-fi calling).

Again, thank you for a good run with the Google Voice Gateway.

Sincerely,
Bill Simon
Simon Telephonics
Bill didn't mention that efforts are underway to retool Asterisk 13 and Incredible PBX 13-13 to support the new Google Voice protocol. You can read this week's Nerd Vittles article for tips on implementation. We also hope to have the Raspberry Pi 3 implementation ready for the pioneers later today or tomorrow.

For those using Google Voice with Issabel, VitalPBX, or 3CX, it's time to start making alternative arrangements to meet your communications requirements.
 

mainenotarynet

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Yes - I got the email too, just a few ago. Reading up on the new implementation, I have a question.

I have the Motif, but since I needed to rebuild my 13-13 anyways (and did soomething completely stupid, like being in the wrong cloud provider's portal and destroying the working one) I built with the Google OAUTH, but have NOT done any of the 4 GV accounts yet. SO the question is since there are no accounts yet does any of the steps change or go with what is in the other thread about this for upgrading the existing systems. We don't USE GV much, but it's nice to have. The REASON we don't is it forces it's TRUE number on CID and not our compartmentalized ones we WANT people to have. Would be great if maybe their new SIP way would allow for this in that its a great FREE fall-back in case voip.ms ever went down -- Secondly would the current refreshtokens I have saved locally work in this new implementation or do I need to get new ones.

Thanks Mr Mundy.
 

wardmundy

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@mainenotarynet: You should be good to go as long as it's a CentOS 6, 64-bit platform. Limitation currently is you can only use one GV account, but I'm working on it. Your current refresh tokens will work as is. Good luck. Let us know how it goes.
 

mainenotarynet

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Yes have CentOS6.9 as when IO built them CentOS7 wasn't ready for 'prime-time' - I have 4 but to see how it goes one will do. my setup is a bit convoluted anyways (Maine Notary Net / Kitty Couch Crafters / Pine State Amateur Radio Club / Revelation 3:11 Ministries) all have GV numbers but also have numbers on voip.ms so GVoice is not critical but since it FORCES its own GV number as the CID, and I want to keep these private [maybe as a backend DISA for outside, possibly an amateur Radio Autopatch substitute -- working on that later] I don't have them in outbound routes for that reason.

Let us know when more than one can be used (481, 482, 483, etc ate the prefixes I use for each number but that limits to 9 (or 10 if zero is used too).

I have to rebuild two Incredibles, I remember a post on two PBXes where a third 'witness server' will verify Main is down and switch to the secondary, but that is a bit much unless the secondary asterisk is activated/deactivated at that time as trunks would need to rergister on secondary until primary is back up. WAZO had this but I couldn't get the hang of it.
 

Robert Johnson

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Ward, currently utilizing the scientific linux 6.9 ESXI VM build, awesome work and love this have had it running for a while now, always updating though.Does the GV Motif use XMPP Interop that is built into Incredible PBX? I have several accounts on there now , everything is still working good, but I am unsure if I will need to posture for a new build. Thanks
 

wardmundy

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You're OK for today, but nobody knows when Google will pull the plug. If you're on CentOS or SL 6.9, the change is pretty easy with the tutorial. Just remember to remove your existing XMPP trunks in FreePBX and restart Asterisk BEFORE you start the upgrade.
 

restamp

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Ward, are the GVsip modifications to 13-13 considered final, or is NAF still resolving issues with them? (Over at dslreports, it looks like the latter.) If they are indeed still being worked on, how will 13-13 be kept current? Via updates?

FWIW, a few days ago I reloaded what has become my test VPS with 13-13 and applied gvsip-naf to it. The Asterisk rebuild went smoothly, but the GVsip trunk still exhibited some glitches. Nothing failed consistently, and thus I didn't do a report on it because I couldn't tell if they were legitimate GVsip issues or ancillary problems due to my testing paradigm, like the GV spam trap problem you described elsewhere. Has anyone else conducted any testing on the NAF modifications?

In any event, with the announced impending demise of the GVGW, when XMPP is finally turned off I intend to move my GVGW trunks over to NAF's software, but if he's fixed some issues in the interim it would be nice to go with the latest and greatest.
 

Eliad

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Does anybody know how many concurrent outgoing calls are we allowed to use on a GV trunk? Is it different with NAF software?
 

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