GVoice In Flames No More

johnnypuffs

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I remember you asked the question and I recall replying but don't remember what I said. I may have noted that so far it seems that only Asterisk 1.8 users are experiencing this issue, and from what I've heard it is affecting neither FreeSWITCH nor Obihai device users (I recall saying that somewhere, but not sure if it was here). Other than that, I know I posted a reply but have no idea what I said. Of course, most days I'm lucky if I can remember what I had for dinner last night! :confused5:

I will guarantee you that I didn't remove my reply, and I don't have the ability to remove your posts.

Thanks MT. Actually, my question was just part of a post.. and it was not removed..

JP
 

johnnypuffs

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Seems to me there was a post that started off in another direction on another topic. The attempt to move it didn't go well. Sorry. :biggrin5:

MORAL: Start new topics in new threads! If your sentence begins with "on a different subject," that's a big hint. :smile5:

Hey Ward,

I think you are referring to me here.. not sure. I don't see any other replies that were pulled, except the one to my question.

My question, which is still there to see, didn't start with "on a different subject"

I asked as part of a question:

Quick question.. I've wondered this for a while but never quite figured it out.. Is Google TRYING to break gvoice for people using asterisk or are these simply "upgrade" on their part that just happen to screw us all..

This thread is about GV breaking yet again. I think that question is 100% on topic. If the reason GV keeps breaking is because Google is purposely doing so it would entirely change my (and I would think others) approach to the situation.

If they are just bugs, I can live with that. If it's a cat and mouse with Google.. (ie. they fix it, we break it again), then it's time for me to look for a better solution..

Again, as I said to MT, I really meant is Google trying to stop people with ANY outside system from using GV..

Thanks,
JP
 

tiggerpaws

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Also increased to 8 seconds in the wait
line and it works now.

My question is, why would goog want to
break it on purpose?

What gain cood they get from keeping
us tied to the computer?

I keep breaking the tie, and therefore
breaking google.
 
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I doubt that Google is intentionally breaking access to voice just via Asterisk and not other similar platforms. However not knowing Google's master plan for Voice which I would assume involves either revenue generation (eventually charging for Voice or partnering with carriers to integrate the service like the recent deal with Sprint.) Or revenue protection some future integration with adwords or search. Who knows if the recent changes were a direct change to make voice users only use Google authorized connection interfaces or a fix of some problems voice users were experiencing on other platforms.

That being said....Google makes it pretty clear that they are not real fond of people accessing their services by means other than through the interface provided by Google.:rolleyes5:


5.3 You agree not to access (or attempt to access) any of the Services by any means other than through the interface that is provided by Google, unless you have been specifically allowed to do so in a separate agreement with Google. You specifically agree not to access (or attempt to access) any of the Services through any automated means (including use of scripts or web crawlers) and shall ensure that you comply with the instructions set out in any robots.txt file present on the Services.


15. LIMITATION OF LIABILITY

15.1 SUBJECT TO OVERALL PROVISION IN PARAGRAPH 14.1 ABOVE, YOU EXPRESSLY UNDERSTAND AND AGREE THAT GOOGLE, ITS SUBSIDIARIES AND AFFILIATES, AND ITS LICENSORS SHALL NOT BE LIABLE TO YOU FOR:

(A) ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL CONSEQUENTIAL OR EXEMPLARY DAMAGES WHICH MAY BE INCURRED BY YOU, HOWEVER CAUSED AND UNDER ANY THEORY OF LIABILITY.. THIS SHALL INCLUDE, BUT NOT BE LIMITED TO, ANY LOSS OF PROFIT (WHETHER INCURRED DIRECTLY OR INDIRECTLY), ANY LOSS OF GOODWILL OR BUSINESS REPUTATION, ANY LOSS OF DATA SUFFERED, COST OF PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES, OR OTHER INTANGIBLE LOSS;


(II) ANY CHANGES WHICH GOOGLE MAY MAKE TO THE SERVICES, OR FOR ANY PERMANENT OR TEMPORARY CESSATION IN THE PROVISION OF THE SERVICES (OR ANY FEATURES WITHIN THE SERVICES);
 

joeg

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Still don't work - anybody have another suggestion?

My PIAF 1.8.3 went on the Google flotch at the same time as everybody else. I made both the Wait(8) and the DTMF(11) fixes and it's still not answering the inbound GV numbers. I have rebooted the system after every edit.

Arrrgghhh :banghead:

Here is my googlein stuff from extensions_custom.conf. Please help!

[googlein]
exten => _.,1,GotoIf($["${EXTEN}" = "[email protected]"]?googlein1,s,1)
exten => _.,n,GotoIf($["${EXTEN}" = "[email protected]"]?googlein2,s,1)

[googlein1]
exten => [email protected],1,Wait(1)
exten => [email protected],n,Set([email protected])
exten => [email protected],n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(name)=${CALLERID(number)})
exten => [email protected],n,GotoIf(${DB_EXISTS(gv_dialout_nrrjoeg/channel)}?bridged)
exten => [email protected],n,Goto(s,regcall)
exten => [email protected],n(bridged),Bridge(${DB_DELETE(gv_dialout_ gv-address-1/channel)}, p)
exten => s,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => s,n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(name)=${CALLERID(number)})
exten => s,n(regcall),Answer
exten => s,1,Wait(8)
exten => s,n,SendDTMF(11)
;exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD:)1))
exten => s,n,Goto(from-trunk,gv-incoming,1)

[googlein2]
exten => [email protected],1,Wait(1)
exten => [email protected],n,Set([email protected])
exten => [email protected],n,JABBERSend(asterisk2,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(name)=${CALLERID(number)})
exten => [email protected],n,GotoIf(${DB_EXISTS(gv_dialout_naturesrite2/channel)}?bridged)
exten => [email protected],n,Goto(s,regcall)
exten => [email protected],n(bridged),Bridge(${DB_DELETE(gv_dialout_gv-address-2/channel)}, p)
exten => s,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk2,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => s,n,JABBERSend(asterisk2,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(name)=${CALLERID(number)})
exten => s,n(regcall),Answer
exten => s,1,Wait(8)
exten => s,n,SendDTMF(11)
exten => s,n,Goto(from-trunk,gv-incoming2,1)
 

ohrass

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i have a multi GV Purple Setup but my config looks different on the back end:

Jabber


[asterisk]
type=client ;connection
serverhost=talk.google.com ;route to server
[email protected]/Talk ;change to your username
secret=xxxxxxxx ;change to your password
port=5222 ;
usetls=yes ;
usesasl=yes ;
status=Available ;
statusmessage="Incredible PBX" ;
timeout=100 ;
keepalive=yes

[asterisk2]
type=client ;connection
serverhost=talk.google.com ;route to server
[email protected]/Talk ;change to your username
secret=xxxxxxx ;change to your password
port=5222 ;
usetls=yes ;
usesasl=yes ;
status=Available ;
statusmessage="Incredible PBX" ;
timeout=100 ;
keepalive=yes



extensions_custom conf

[googlein]
exten => [email protected],1,Wait(1)
exten => [email protected],n,Set([email protected])
exten => [email protected],n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(name)=${CALLERID(number)})
exten => [email protected],n,GotoIf(${DB_EXISTS(gv_dialout_xxxxxx/channel)}?bridged)
exten => [email protected],n,Goto(s,regcall)
exten => [email protected],n(bridged),Bridge(${DB_DELETE(gv_dialout_xxxxxx/channel)}, p)
exten => [email protected],1,Wait(1)
exten => [email protected],n,Set([email protected])
exten => [email protected],n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => [email protected],n,Set(CALLERID(name)=${CALLERID(number)})
exten => [email protected],n,GotoIf(${DB_EXISTS(gv_dialout_xxxxxx/channel)}?bridged)
exten => [email protected],n,Goto(s,regcall)
exten => [email protected],n(bridged),Bridge(${DB_DELETE(gv_dialout_xxxxxx/channel)}, p)
exten => s,1,Wait(1)
exten => s,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => s,n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(name)=${CALLERID(number)})
exten => s,n(regcall),Answer
exten => s,n,Wait(8)
exten => s,n,SendDTMF(11)
exten => s,n,Goto(from-trunk,gv-incoming-${CUT(ALERTNAME,@,1)},1)

[gvoice]
exten => _X.,1,Wait(1)
exten => _X.,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => _X.,n,JABBERSend(asterisk,${ALERTNAME},Placing GVoice Call: ${EXTEN})
exten => _X.,n,GotoIf($["${LEN(${EXTEN})}"="10"]?us:eek:ther)
exten => _X.,n(us),Dial(Gtalk/asterisk/+1${EXTEN}@voice.google.com)
exten => _X.,n,Goto(done)
exten => _X.,n(other),Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
exten => _X.,n(done),NoOp(GVoice Call to ${EXTEN} failed)

[gvoice2]
exten => _X.,1,Wait(1)
exten => _X.,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk2,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => _X.,n,JABBERSend(asterisk2,${ALERTNAME},Placing GVoice Call: ${EXTEN})
exten => _X.,n,GotoIf($["${LEN(${EXTEN})}"="10"]?us:eek:ther)
exten => _X.,n(us),Dial(Gtalk/asterisk2/+1${EXTEN}@voice.google.com)
exten => _X.,n,Goto(done)
exten => _X.,n(other),Dial(Gtalk/asterisk2/+${EXTEN}@voice.google.com)
exten => _X.,n(done),NoOp(GVoice Call to ${EXTEN} failed)
 

gvtricks

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This is my setup and is working fine now.
VMwarePlayer on Nat network (this allow me to go Mobil with my laptop with a fix IP 192.168.121.136 so I dont have to change anything in my free version of XLite)

PBX in a flash = 1.7.5.5
FreePBX version = 2.8.1.4
Running Asterisk Version = 1.8.3.2
Asterisk Source Version = 1.8.3.2
IP address = 192.168.121.136

No ports forwarding at all.

I am also running HylaFAX Avantfax from the following setup

[URL="http://pbxinaflash.com/community/threads/hylafax-avantfax-for-piaf-parts-1-4.3645/?t=3645"[/URL]

and is working great for sending faxes with Google voice, I have been able to send 20 pages .pdf files with no problem.

My GV-Incoming Route have been setup for 701 extension Destination.

exten => s,1,Wait(1)
exten => s,n,Set([email protected])
;exten => _X.,n,Set(STATUS=${JABBER_STATUS(asterisk,${ALERTNAME})});
;exten => _X.,n,NoOp(Gvoice/Jabber Status: ${STATUS})
exten => s,n,JABBERSend(asterisk,${ALERTNAME},Incoming Google Voice Call: ${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(number)=${CALLERID(name):2:10})
exten => s,n,Set(CALLERID(name)=${CALLERID(number)})
;exten => s,n(regcall),Answer
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD:)1))
exten => s,n,Wait(3)
exten => s,n,SendDTMF(11)
exten => s,n,Goto(from-trunk,gv-incoming,1)

Yes I was having the same problem with gvoice incoming and I don have to press a one when I pickup now.

I also like to thank Ward, Tom and every body else contributing to this Forum.

If this can help any one

Good luck

gvtricks
 
Last edited by a moderator:

wardmundy

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FWIW Dept.

That being said....Google makes it pretty clear that they are not real fond of people accessing their services by means other than through the interface provided by Google.:rolleyes5:


5.3 You agree not to access (or attempt to access) any of the Services by any means other than through the interface that is provided by Google, unless you have been specifically allowed to do so in a separate agreement with Google. You specifically agree not to access (or attempt to access) any of the Services through any automated means (including use of scripts or web crawlers) and shall ensure that you comply with the instructions set out in any robots.txt file present on the Services.


Google provides the Gtalk interface to its service. We use the Gtalk interface provided by Asterisk to access Google Voice. Case closed. :innocentb:
 

amygrant

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There seems to be a timing quirk that's been injected at Google's end. For those using Incredible PBX, new downloads already have been patched. For existing installs, here's the fix that seems to work. In /etc/asterisk, edit extensions_custom.conf. Scroll down toward the end of the file to the [googlein] context. Make the last 5 lines of the context (beginning with the regcall Answer line) look like this and then restart Asterisk:

Thank you Ward. Did that on 3 servers and it fixed them all. Charlie Sheen has nothing on you!
 

wardmundy

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***WINNING***

images
 

amygrant

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Wanted to post an update on this.

I implemented the 8 second fix back when this issue first arose, and it worked like a champ.

Today suddenly I started having the same problem on 3 different servers. I upped the wait time to 10 seconds and it fixed it on all 3.
 

voipRookie

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Wanted to post an update on this.

I implemented the 8 second fix back when this issue first arose, and it worked like a champ.

Today suddenly I started having the same problem on 3 different servers. I upped the wait time to 10 seconds and it fixed it on all 3.


I was having the same problem and your fix worked for me also.
 

DerekX

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Same here! The problem had returned where incoming calls went to Google Voice mail after one ring. Applying you fix did the trick.
 
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Here's another possible solution to this mess that MIGHT eliminate the need to add waits. I posted in a previous message how you could use a DID to bring in Google Voice calls. But if for some reason you don't want to do that, but you are still tired of the hassles trying to make this work, I will just note that from all the reports I'm hearing, this is only a problem for Asterisk 1.8 users that are using the Jabber/XMPP to bring incoming calls in. If you can receive the incoming calls without using the Jabber channel driver then in most cases there is no problem (which is why a DID works).

In the last couple of days I've read a couple of different suggestions for bringing in GV calls:

The first is to set up a free account on pbxes.com. Apparently they have offered support for Google Voice since last October, according to this announcement. A reader of my blog named Zack said this:

It's worth repeating that, when you access Google Voice through pbxes.org, the dial-1 problem doesn't exist. It's weird, because pbxes.org seems to use a very Asterisk-like interface (it even lets you see the source code, which is basically Asterisk source), and I always assumed it actually used Asterisk to access Google Voice. I wonder if pbxes.org has some special deal with Google or if their software somehow identifies itself differently to Google Talk or uses a modified jabber client. BTW, a pbxes.org account is free and gives you full access to GV (if you set the account up with an Android phone). It's a good solution to this problem, though you need to use an intermediary.

I haven't tried this yet and if anyone does, I'd be interested to know if it solves the problem. I'd assume that if it works, you could use it as an intermediary between your Asterisk server and Google Voice, and I'd also assume that you would NOT need a DID for this to work.

A second approach would be to get an account with Tropo - see this article and video. The interesting thing about doing that is that you'd also be reachable via a public SIP address (without having to open your firewall to anyone and everyone) and you'd also be reachable from Skype and via an iNum number, in addition to the DID you'd get from Tropo. This approach would still use a PSTN number to receive calls from Google Voice (supplied by Tropo), but as I mentioned, most users don't have any problems when GV sends a call to a PSTN number. The only thing about Tropo is that you can't be sure if you will ever have to pay for their service - right now I believe that a "developer account" is free to anyone who signs up for one. You'd also have to know enough about programming in one of their supported languages to at least be able to transfer calls to your system (via SIP, of course). This one probably isn't for everyone, since there are other ways to get DID's. I had played with Tropo a bit back in December but on the whole I wasn't all that impressed with it, although I can see where programmers might view it much more favorably.
 

woland

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I think pbxes.org works with google voice (although I have not used it for a couple of weeks).
However, there are two issues.

1. The most important at the moment is the fact that they seem to have capacity problems and they disabled sign up for free accounts with GV trunks for the time being. I think they mentioned that they expect to reopen this in May. Also, while not a serious problem, it is worth pointing out that you need a specific sip client, sipdroid, to create free account with GV trunk on adnroid.
2. Even if/when pbxes.org reopens the signup for free accounts, they seem to restrict it to a single account per android phone. So each usable GV phone number requires a cell phone, which is more restrictive than Google requirement for a forwarding number.
 

mpreissner

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I'm having this problem too, but in my extensions_custom.conf, I have a SendDTMF(11) instead of (1). I have never heard the prompt, and pressing 1 immediately when I answer an inbound call is hit or miss whether it works or not...
 

RTOMikey

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I am beginning to think it has something to do with sensing automation. If I send it to a IRV or queue, all my lines work about 3 out of five tries, however, if I change the inbound route to a direct extension, it goes through 100% of the time, following the same wait(8), Dial (11) that I have set up. Perhaps is a new way to defeat using these lines for commercial apps? I know callcentric does the same thing to me with the Dirt Cheap DIDS that I use for some of my customers renting phone service from me. when it goes to their extension it works, when they cancel and I pull it back to my corral and point to general IVR, it stops calls at callcentric saying not available
 
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I'm starting to think it's a timing issue. One thing that most of us do, because we don't want Google Voice to snatch the call back and send it to their voicemail when we have a perfectly good voicemail system of our own, is to answer the call the second it arrives. Google Voice used to be okay with that, but then I had one case where calls were coming into a particular Google Voice number which was sending them to a DID coming into the system, and if I did the standard "answer at connect" the calls didn't work about 75% of the time — specifically, they would come into the system and ring the called extension, but if the called party answered they'd get silence while the caller would just hear ringing until the call went to GV voicemail. The trick seems to be that you have to let the incoming Google Voice call "ring" for a second or two and THEN answer it. So, in your Inbound Route, you can set the "Pause Before Answer" setting to 1 or 2, and then from there send the call to whatever you use to answer it.

Please note that the above worked when Google Voice was sending the call to a DID that comes into the system, NOT when using the Asterisk 1.8 channel drivers to receive the incoming call.

So I'm starting to think the root of this issue is one of two things:

1) Google Voice is getting overloaded and if there is some kind of "race" condition taking place, where if the "answer" comes back before they are ready to receive it, they treat it as if the call is unanswered. Note that there might be enough delay inherent when the call goes through on certain DIDs that you'd never have a problem, while other, faster DID's might run afoul of this.

2) They have actually implemented some kind of code that deliberately disregards destinations where the call is answered "too soon", that is, before the first ring. Who knows why they might do something like that — your guess is as good as mine.

Now, as I think I mentioned before (too lazy to go back and read my previous posts), the problems we are having with incoming calls ONLY seem to affect Asterisk 1.8 users. And I know that Malcolm Davenport at Digium was trying to find a solution for this and then all the tornadoes hit, knocking out utilities and causing major issues for Digium employees, so whatever he was working on has probably been delayed at least a couple of weeks, if not longer. Hopefully, eventually, the Asterisk channel drivers will be fixed and then we can go back to using those or take out all the waits or whatever we've done to try to work around this issue in the meantime.
 

chemcat9

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Just wondering the status of this, I have a fresh install of 1.8 Purple exten => s,n,Wait(8) and am seeing this behavior. Odd how my extensions would initially ring, and now it rings my cell and then fails over to google vm.
 

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