FYI An option for keeping Google Voice in the post-XMPP Era

[restamp]

It appears that there are two groups of people here: The first consists of entrepreneurs, configuring, maintaining, and providing VoIP services to corporate clients. The second consists of hobbyists like me who like to tinker in exchange for getting voice calling for free, or nearly so. I suspect this post will mainly be of interest to those in the latter group.

Many of us hobbyists rely on Google Voice for our trunking needs. It's certainly not perfect, but it can be made to work reliably. Recently, Google announced it will be shutting down its XMPP servers -- the protocol Asterisk uses to access GV. Polycom (neé OBiHAI) makes the only non-Google devices which are currently guaranteed to have access to the new GV network.

This got me thinking: The OBi 200 and 202 devices are themselves a feature-rich ATA capable of handling four trunks and/or endpoint devices. Could they be pulled into service to provide a SIP-to-GV bridge? Think of this as a small localized version of Bill Simon's Google Voice Gateway. Initial tests have proved promising.

Without going into the details of how to setup these devices, 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 it would bridge them to a corresponding GV line to complete the call. And (2) incoming GV calls would be automatically translated into SIP and delivered to Asterisk. In essence, the addition of an OBi200 or 202 ATA to an Asterisk server is all that it should theoretically take to continue to support *two* GV lines as Asterisk trunks.

Is this a panacea? Of course not. It does nothing to address the problem of retaining GV trunking on an Asterisk VPS. It should also be noted that this experiment takes place using OBi boxes which still employ the old GV XMPP protocol. There are no guarantees that those with the new protocol will continue to work identically. But it does give us another potential option to use in the unlikely event that no workaround is developed for interfacing Asterisk to the new GV protocol and the Google Voice Gateway shuts down.

If anyone would like to see all the nitty-gritty details for configuring and interfacing the OBi boxes to Asterisk, just speak up and I'll be happy to provide them.

Edit: See the following URL for all the "nitty-gritty details": https://www.dslreports.com/forum/r31954649-

 

[tycho]

There are many who will surely want to see the nitty-gritty details. For my part, I will probably defer, waiting to see if Obi actually does end up being the only game in town. Clever minds loving a challenge, it may well come to pass that others can "find a way in." If not, for my limited personal hobby needs I'd probably just hook-up the Obi 200 that I bought a while ago on deep sale and that is still in shrink-wrap, and be done with it.

Despite my many PBXs, both local and in the cloud (usually 2 of each at any given moment in time), my outbound calling needs are stupid simple. Quite a while ago, during an early iteration of the "Google Broke the Obi" games, I migrated to using the Simonics GVGW on "SP1" on my old Obi 100 for home use. I pick up line one of my cordless analog DECT phone and make a call without drama. By using codes I can also access my a friend's PBX; my own local HW PBX; and 5 VoIP providers -- it is a handy little box.

So, if that ability breaks and Bill is shut-out, I'll swap-out the Obi 100 for the Obi 200. Unless someone comes up with something clever, I'm likely to keep GV out of the PBX zone...

 

[billsimon]

So, if that ability breaks and Bill is shut-out

We'll see. Making good progress so far!

 

[restamp]

We'll see. Making good progress so far!

Great, Bill! A revised working Gateway would of course be my clear preference. Could you tell us whether Google is working with you (providing you with the technical details of their new protocol) or not?

(An interesting side note: While testing incoming calls with the OBi setup above, I noticed the same 1-to-2 second delay in bridging the audio through after answering that has become a signature of the GVGW.)

 

[billsimon]

Great, Bill! A revised working Gateway would of course be my clear preference. Could you tell us whether Google is working with you (providing you with the technical details of their new protocol) or not?

They are not. I'm just getting my software ready for when they call.

PS: no more 1-2 second delays.

 

[tycho]

They are not. I'm just getting my software ready for when they call.

And you still believe in Santa Clause?

PS: no more 1-2 second delays.

This is great, although I long ago trained myself to not even notice it!

 

[yajrendrag]

@restamp - i followed your "nitty-gritty details" but can't make outgoing calls work - incoming work fine. I had an obi202 laying around from doing this in the past, so reset it to factory and replicated your SP1 and SP4 settings.

On outgoing calls from asterisk (also a 13-13 system) I get this error:

Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

I'm calling another asterisk system and the error is from the originating end - call never makes it to destination. error seems to be related to originating asterisk trying to call through obi...

Any suggestions?

running obi f/w version: 3.2.2 (Build: 5859EX) and have also unchecked X_EnforceRequestUserID

thx, jay

 

[restamp]

Jay, although I managed to get it working here, I don't consider myself an expert on the OBi or Asterisk by any stretch of the imagination. (Thus, if anyone has any suggestions, feel free to interject them.)

I'm guessing the problem is that Asterisk and the OBi are not talking. I'm further presuming that both the A* server and the OBi are on the same local LAN (i.e., no firewall between them).

If you probe A*, with "asterisk -x 'sip show peers' | grep -v Unmonitored", does it show the OBi trunk as "OK" with a small RTT? If not, the two aren't talking. You say you are using SP1 for your A*-OBi sip connection, right? If so the port 5060 should work by default, but verify that the SP1 X_UserAgentPort is set to 5060 on your OBi and none of the other SPx X_UserAgentPorts conflict. If none of the above is abnormal, let me know and I'll give it some more thought.

Good luck.

 

[yajrendrag]

SOLVED.
@restamp - thanks very much. that X_UserAgentPort setting was the problem. it was set to 5080. i stared at that for a few minutes last night while i was fiddling with this, but assumed it was correct after resetting the device to the factory settings. Anyway, after making that correction, the "asterisk -x 'sip show peers' | grep -v Unmonitored" command did indeed show OBi trunk as OK with small RTT... prior to that change it was unreachable. & now outgoing calls work as well.

 

[Eliad]

how many goggle voice I can configure on one Obi 200, it seems to be able to support 4 google accounts. I am not sure if i can still do 4 google accounts when I connect it to incrediblePBX FreePBX version. Did anybody try more than one google account?

 

[restamp]

Eliad, the following post may prove helpful:

https://www.dslreports.com/forum/r31956002-