SOLVED GV stopped connecting today

NIkesh Shakya

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Yeah

I am having the issue too

[2016-06-23 10:59:00] WARNING[1831]: res_xmpp.c:3569 xmpp_client_receive: Parsing failure: Hook returned an error.
[2016-06-23 10:59:00] WARNING[1831]: res_xmpp.c:3633 xmpp_client_thread: JABBER: socket read error
[2016-06-23 10:59:00] ERROR[1829]: res_xmpp.c:2556 xmpp_client_requested_tls: TLS connection for client 'emailgmailcom' cannot be established. OpenSSL initialization failed.
 

billsimon

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There are no complaints on the Obihai forum today. It looks like the updates they did last time have gotten them through today's situation.
 

Erik Knowles

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Chiming in with my own $0.02...

I don't use Google Voice, but I do use the XMPP client to pop up incoming call notifications. This morning I came into work to find my BeagleBone FreePBX server locked solid (after 16 months uptime). Rebooting shows a solid stream of res_xmpp.c:2556 xmpp_client_requested_tls: TLS connection for client 'erik' cannot be established. OpenSSL initialization failed errors, and the server is now showing multiple VOIP trunk connection lossses in the last hour, presumably due to the constant XMPP connection attempts filling up a table somewhere.

So, yeah, bad design if an offline chat server can bring down the whole voip system.
 

Erik Knowles

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Update -- the problem remains after updating the core (I was one version behind).
 

billsimon

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Special offer for NerdVittles / PBX in a Flash Forum readers: (with Ward's blessing to post this promotion)

$1 off Google Voice Gateway (GV-to-SIP) service using this link: https://simonics.com/gw?promo=nerdvittles (your price becomes $4.99)

Read more about the service at http://simonics.com/services/

Google Voice Gateway has been reliable through the recent disruptions of GV service to Asterisk users. I understand that most participants in this forum are a tenacious, sometimes hacker-like crew that will work through these issues and come out stronger after the fact. But if you just want it to work using a standard SIP trunk setup instead, it only costs $4.99 (one-time) -- offer limited to readers of this post.

Once you set up a Google Voice Gateway account, you can follow this guide to set up your Incredible / PIAF / FreePBX trunk.
 

rob-pbiaf

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As noted earlier in this thread, I am using an older asterisk embedded within my router, and I again suffered a router crash due to this problem. I already determined last time that there must be a design flaw in the Motif/Xmpp implementation within asterisk that losing one trunk can crash the asterisk process and hang the Linux-based host the way it did.

I have today jumped at the chance to convert my configuration to replacing the direct Google link with one passing through the Google Voice Gateway where I trust that the SIP functionality is sufficiently robust that it won't crash asterisk if for some reason the gateway's presentation of GV goes down. And I will be quite happy to stop seeing all the Xmpp presence warnings of announcements not on my roster come through my error log.

I also get the added benefit that nowhere in my setup is my GV plain text password stored in my own configuration files or in my entries to the GVGW sign-up process. That is a great use of the "sign in with Google" feature. Getting the CID-Name feature of GVGW is an added bonus.

I do hope that the direct jabber configuration will again work for me, and that asterisk will become more impervious to these types of outages, but I'm a happy GVGW customer protected from this type of repeat outage.


Special offer for NerdVittles / PBX in a Flash Forum readers: (with Ward's blessing to post this promotion)

$1 off Google Voice Gateway (GV-to-SIP) service using this link: https://simonics.com/gw?promo=nerdvittles (your price becomes $4.99)

Read more about the service at http://simonics.com/services/

Google Voice Gateway has been reliable through the recent disruptions of GV service to Asterisk users. I understand that most participants in this forum are a tenacious, sometimes hacker-like crew that will work through these issues and come out stronger after the fact. But if you just want it to work using a standard SIP trunk setup instead, it only costs $4.99 (one-time) -- offer limited to readers of this post.

Once you set up a Google Voice Gateway account, you can follow this guide to set up your Incredible / PIAF / FreePBX trunk.
 

JRoque

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Google Voice Gateway has been reliable through the recent disruptions of GV service to Asterisk users. I understand that most participants in this forum are a tenacious, sometimes hacker-like crew that will work through these issues and come out stronger after the fact. But if you just want it to work using a standard SIP trunk setup instead, it only costs $4.99 (one-time) -- offer limited to readers of this post.

Hi! Have you fixed this issue? What guarantees do you provide that this issue, which appears to be initiated at Google's end, won't happen to your service?

JR
 

billsimon

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Hi! Have you fixed this issue? What guarantees do you provide that this issue, which appears to be initiated at Google's end, won't happen to your service?

We never encountered the issue, so there was nothing to fix. I can't offer any guarantees.

Google Voice Gateway uses OAUTH2 token-based login to Google Voice and the most current OpenSSL libraries. Also, it does not use Asterisk. Thus we are clear of some of the areas folks have suspected as being the problem (plain login instead of OAUTH2, outdated TLS methods, Asterisk chan_motif bug).
 

wardmundy

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Hi! Have you fixed this issue? What guarantees do you provide that this issue, which appears to be initiated at Google's end, won't happen to your service?

JR

Just curious what sort of guarantee you would expect on a $4.99 purchase of a lifetime SIP connection to a service created and maintained solely by Google? :crazy:
 

VaHam

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My old hardware version of PIAF has failed to register with GV also however my newer version of PIAF (without OAuth) remains working fine today. My Obi110's also failed because I had not performed the latest updates to them; however after updating them they are indeed working fine also. http://www.obitalk.com/forum/index.php?topic=9.0

old-piaf.png

NOT WORKING

24Jun2016-CaC.png

WORKING

I agree with Ward !!!!! The $4.95 to Bill is well worth it. I never have problems with those trunks. :clap:
 

JRoque

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Just curious what sort of guarantee you would expect on a $4.99 purchase of a lifetime SIP connection to a service created and maintained solely by Google? :crazy:

The kind of guarantee that would be implied when someone posts into an active issue thread to offer a commercial solution. I just thought it prudent to ask what was done to solve the issue affecting many of us.

JR
 

billsimon

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The kind of guarantee that would be implied when someone posts into an active issue thread to offer a commercial solution. I just thought it prudent to ask what was done to solve the issue affecting many of us.

I forgot to mention that there is a 30 day refund guarantee (<-- link). Since it seems this issue is now appearing weekly for Asterisk users, that would give you a few weeks to see for yourself whether Google Voice Gateway works well for you. If not you can request a refund.
 

mrfeh

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Best $4.99 you'll ever spend. Seriously!

I'm certainly willing to spend the money, but is there a guide anywhere for what needs to be done to migrate from incrediblepbx to the method used when using Google Voice Gateway?

A little quick background - I came to the world of running my own PBX just a few weeks ago. I'm not a telephony engineer, so some of the terms/concepts are still greek to me. I managed to get everything working by following this guide:

http://nerdvittles.com/?p=10151

I have an ATA and a Raspberry Pi B+. Everything was working...got home from vacation yesterday and it now appears I cannot get the trunk connected to Google Voice. It seems to be the issue this thread covers.

What do I need to turn off or uninstall from my incrediblepbx installation? Do I need to install a different image?

Any pointers would be helpful. Thanks!
 

mrfeh

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I have my GVGW account set up, and the GVGW account page shows "devices registered: 1", so I think the SIP trunk is set up.

However, incoming calls are not (apparently) reaching my home, and if I try to place an outgoing call, I get a recording that says I can't place the call as dialed.

I've followed the instructions on this page: http://nerdvittles.com/?p=12807

But, still no go. How do I troubleshoot?
 

Kennybell

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I have my GVGW account set up, and the GVGW account page shows "devices registered: 1", so I think the SIP trunk is set up.

However, incoming calls are not (apparently) reaching my home, and if I try to place an outgoing call, I get a recording that says I can't place the call as dialed.

I've followed the instructions on this page: http://nerdvittles.com/?p=12807

But, still no go. How do I troubleshoot?

Make sure you have inbound and outbound routes for the trunk.

I just setup mine to replace the motif and all works fine. (my inbound & outbound routes were already there)
 

tenortim

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Heh, I should have looked for this thread before spending a few hours debugging the problem(s)!

Initially, I went ahead and patched to support oauth2, but it didn't help.
I then dived into the res/res_xmpp.c code and added the error checking and handling that should have been there in the first place and that's where I discovered the problem that has already been discussed in this thread:
I was/am running Asterisk 11.10.2, and the version of res_xmpp.c in that version of Asterisk is hardcoded to only allow SSLv3.
It seems Google just turned off SSLv3 support, so only TLS is supported.

So, if you're on Asterisk 11 or earlier, you need to either upgrade to a later version of, at a minimum, Asterisk 11, or patch res_xmpp.c and at a minimum, change the initialization of ssl_method from
SSLv3_method()
to
SSLv23_method()
If you do that, you really ought to add the extra two lines from later releases that block SSLv2/v3 viz:
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
SSL_CTX_set_options(client->ssl_context, ssl_opts);
(and declare ssl_opts as a long back at the beginning of the function).

Upshot, I'm still running 11.10.2 (however briefly) and once again, my GV is working. I think this is a hint for me to upgrade :)

[Edit]Clarifications
RPI-Fan and billsimon called it.
At least in my case, and probably others, Google have stopped accepting SSLv3 connections.
The older chan_xmpp.c code was hardcoded to only use only SSLv3.
The function names are less than obvious:
SSLv3_method() locks down your connection to only use SSLv3.
SSLv23_method() actually negotiates, choosing the highest common method, and If you're running at least OpenSSL v1.0.1, that includes all of TLSv1, TLSv1.1, and TLSv1.2.
 
Last edited:

Ivan Apolonio

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Heh, I should have looked for this thread before spending a few hours debugging the problem(s)!

Initially, I went ahead and patched to support oauth2, but it didn't help.
I then dived into the res/res_xmpp.c code and added the error checking and handling that should have been there in the first place and that's where I discovered the problem that has already been discussed in this thread:
I was/am running Asterisk 11.10.2, and the version of res_xmpp.c in that version of Asterisk is hardcoded to only allow SSLv3.
It seems Google just turned off SSLv3 support, so only TLS is supported.

So, if you're on Asterisk 11 or earlier, you need to either upgrade to a later version of, at a minimum, Asterisk 11, or patch res_xmpp.c and at a minimum, change the initialization of ssl_method from
SSLv3_method()
to
SSLv23_method()
If you do that, you really ought to add the extra two lines from later releases that block SSLv2/v3 viz:
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
SSL_CTX_set_options(client->ssl_context, ssl_opts);
(and declare ssl_opts as a long back at the beginning of the function).

Upshot, I'm still running 11.10.2 (however briefly) and once again, my GV is working. I think this is a hint for me to upgrade :)

[Edit]Clarifications
RPI-Fan and billsimon called it.
At least in my case, and probably others, Google have stopped accepting SSLv3 connections.
The older chan_xmpp.c code was hardcoded to only use only SSLv3.
The function names are less than obvious:
SSLv3_method() locks down your connection to only use SSLv3.
SSLv23_method() actually negotiates, choosing the highest common method, and If you're running at least OpenSSL v1.0.1, that includes all of TLSv1, TLSv1.1, and TLSv1.2.

Man, you saved my life!!

I was desperate because my Google Voice stopped working yesterday and I was following this thread anxiously hopping someone could give me a light about how to workaround this issue. Finally I've read your post and thought: it's a too simple solution to doesn't try and it just worked like a charm! Just edit res_xmpp.c and replace SSLv3_method() to SSLv23_method() and recompile asterisk (actually it just recompiled this module). By simply doing this, my Google Voice is now working again!

asterisk*CLI> xmpp show connections

Jabber Users and their status:

[google] [email protected] - Connected

----

Number of clients: 1

Thanks again!!
 

wardmundy

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Thanks, @tenortim. On really old systems, you'll need to make the SSLv3_method() to SSLv23_method() change in res/res_jabber.c instead of res_xmpp.c.
 
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