ALERT GV: The Sky Has Fallen... Really

[Waffull]

I ran this on my Ras3B+ w/ RASPBX. During the install I noticed this error:

grep: /etc/asterisk/pjsip_custom.conf: No such file or directory

After the process completed, I checked the legacy google voice and didn't see a new obitalk device. So created another refresh token and added gvsip2... no obitalk device registered with google voice. I took the 2nd refresh token and added gvsip2 (non-gui version - 7dad968) on my production server which is IncrediblePBX (13.0.195.5/13.21.1,) which registered it with google voice without a problem.

So I'm not sure if something went wrong with the compilation when I first isntalled this gui version or what. I can send some log files later when I get back to my computer... just wanted to give you a heads up in case this makes any sense.

Thanks.

GVSIP-NAF-GUI (beta)

For the pioneers (only), we've got a new GVSIP-NAF implementation specifically tailored to management within the FreePBX GUI. It should work with CentOS/SL, Debian/Ubuntu, and Raspbian. Works better with Incredible PBX because we know what's been installed.

Translation: You are responsible for creation of Inbound and Outbound Routes to manage your GVSIP trunks, numbered 1-n. The installer will handle creation of the GVSIP trunks themselves. It takes about 10 seconds to add a new GVSIP trunk. You can add as many as you like.

Overview: Running the installer (install-gvsip) the first time will get your Incredible PBX platform up to speed by installing the correct version of OpenSSL for your platform. Then it installs and patches Asterisk 13.21.1 to support GVSIP Google Voice trunks. Finally, it will let you create GVSIP trunks by simply entering a refresh_token and 10-digit phone number for your existing Google Voice trunk. For each trunk, the installer will create the necessary code to support a PJSIP trunk and a GVSIPn Custom Trunk to use for outbound routing. You can run the installer multiple times without worry. The second time you run it, you can install GVSIP trunk #2. The third time, it's GVSIP trunk #3. There's no limit. You can delete existing GVSIP trunks (#1 through #9) by running del-trunk. If you need to delete trunks higher than 9, edit del-trunk and add new sections for the number of trunks you have.

Setup: Once you have added at least one GVSIP trunk, you will need to go into FreePBX with a browser and add an Outbound Route for each of your trunks. Outbound calling with your new trunks will not work until you do this. We recommend dialing prefixes of *41-*49 for outgoing calls, but you can set things up however you like. That's what the GUI is for. Incoming calls to new trunks by default will go to Allison's Demo IVR if you're using Incredible PBX. You should add an Inbound Route for each of your trunks using the 10-digit DID of the GV trunk and specifying a destination for the incoming calls. For non-Incredible PBX users, inbound calls won't be processed until you add an incoming route.

Installation: To get started, login to your Linux CLI as root. Make a backup. Be sure your MySQL root password is set to passw0rd (with a zero) before proceeding. Then...

cd /root
wget http://incrediblepbx.com/gvsip-naf-gui.tar.gz
tar zxvf gvsip-naf-gui.tar.gz
rm -f gvsip-naf-gui.tar.gz
cd gvsip-naf
./install-gvsip

For instructions on obtaining refresh_tokens for your Google Voice trunks, go here. For instructions on creating routes, go here.

Should you ever want to refresh the patched version of Asterisk, copy pjsip_custom.conf to a safe place, delete the contents of pjsip_custom.conf, rerun the installer, and then copy your version of pjsip_custom.conf back to /etc/asterisk and restart Asterisk. That way you won't lose any of your previously configured GVSIP trunks.

 

[kdthomas]

Gave it shot, but didn't work for me. Added #1 Google number and Asterisk DB is down. I rebooted, but still shows "Can not connect to Asterisk". Time to restore from snapshot.

VM running CentOS 7 with latest IncrediblePBX fully patched.

 

[restamp]

GVSIP-NAF-GUI (beta)

For the pioneers (only), we've got a new GVSIP-NAF implementation specifically tailored to management within the FreePBX GUI. It should work with CentOS/SL, Debian/Ubuntu, and Raspbian.

Gave this a try this evening. Only loaded one GVsip trunk. Had to blow away the gvsip outgoing code in extensions_custom.conf which was left over from NAF testing to get outgoing trunks to work. (The incoming code did not seem to present a problem, but I removed that, too.) Bottom line: after creating an inbound and outbound route, everything seemed to work well. I then blew away pjsip_custom.conf, rebuilt everything and verified it all still worked.

A cursory report, but it looks like we have a winner here, thanks to NAF and Ward's hard work (plus all the helping hands testers on the NAF project). Congratulations, folks. Now let's hope Google keeps things stable.

 

[yajrendrag]

will the GVSIP-NAF-GUI make it's way to wazo eventually?

 

[kdthomas]

Did you delete your Google Motif settings and trunks before doing the install? Maybe this is where I went wrong.

 

[wardmundy]

We haven't tested this with CentOS 7 yet. It was a stumbling block in earlier testing of the other releases so that may be an issue. The other thing to check, if you didn't have pjsip_custom.conf on your original server, is whether that file gets built properly and is owned by asterisk:asterisk when the dust settles. Just let me know.

And obviously you need to delete your existing GV trunks from other devices and your servers before beginning.

 

[Waffull]

root@raspbx:/etc/asterisk# ls -l pjsip_*.conf
-rw-r--r-- 1 asterisk asterisk   0 Jul 12 21:10 pjsip_custom.conf
-rw-rw-r-- 1 asterisk asterisk   0 Sep 17  2017 pjsip_custom_orig.conf
-rw-rw-r-- 1 asterisk asterisk   0 Sep 17  2017 pjsip_custom_post.conf
-rw-rw-r-- 1 asterisk asterisk 733 Jul 12 22:19 pjsip_notify.conf

pjsip_custom.conf is empty

 

[wardmundy]

root@raspbx:/etc/asterisk# ls -l pjsip_*.conf
-rw-r--r-- 1 asterisk asterisk   0 Jul 12 21:10 pjsip_custom.conf
-rw-rw-r-- 1 asterisk asterisk   0 Sep 17  2017 pjsip_custom_orig.conf
-rw-rw-r-- 1 asterisk asterisk   0 Sep 17  2017 pjsip_custom_post.conf
-rw-rw-r-- 1 asterisk asterisk 733 Jul 12 22:19 pjsip_notify.conf

pjsip_custom.conf is empty

Very strange. Try running the install again on the existing server and see if that straightens things out. pjsip_custom.conf should never be empty when the install finishes.

 

[Eliad]

Not related. Open a new thread. No fax code is touched by this installer.

Ward
I am not sure about this. I did the installer on a production system and as soon as I installed I got this error. Also, receiving faxes did not work either, it looked it was receiving fax but did not get the fax. I did make a backup of the system prior to the installer and I reversed immediately to the status prior to the installer. After I reversed it I realized I did not check the fax logs, I apologize, I was in a hurry to restore the system.
When I have a chance I will try the installer again and report back to you.

 

[Waffull]

Ran through install again, after deleting the 3 gvsip trunks (using del script,) and then deleting pjsip_custom.conf. At the end, the pjsip_custom.conf file was populated with gvsip1. BUT, the trunk list in freepbx still had a gvsip2 and a gvsip3.

I decided to restore from a week old backup and start from scratch, well at least in regards to gvsip.
Upon running ./install-gvsip, I noticed it now lists Asterisk 13.22.0. From the dslreports thread, it appears that's not a problem, just worth noting so people are aware.

Some other observations:

• The next screen says:

  • This appears to be a fresh install of GVSIP for Incredible PBX.
    Do you wish to install all of the required components? (y/n)

    • Not a big deal, but you may want to change that to GVSIP for FreePBX The way it reads now, is seems to imply choosing yes may install something having to do with IncrediblePBX.
• The window size check misread the size of my window which was actually 121 x 43, but it thought it was 80 x 24.
• After adding refresh token, received this error:

  • We're ready to add GVSIP trunk #1. Press Ctrl-C now to abort.
    Otherwise, press Enter to proceed at your own risk...
    
    Adding GVSIP trunk #1...
    ./install-gvsip: line 422: /etc/pbx/.phone: No such file or directory

• Should the server be rebooted when the script is finished running? Previous versions specified so.
• I checked gVoice legacy page, no new obitalk device added
• After opening FreePBX, this shows up at the top of the page:
  
• I noticed FreePBX wasn't showing the updated version of Asterisk, so I rebooted. Same after reboot, still showing 13.20.0
  

So clearly the install process isn't going as one would expect on a RasPi3B+ with RASPBX (the 4-4-2018 build) w/ FreePBX 14.0.3.6.

Very strange. Try running the install again on the existing server and see if that straightens things out. pjsip_custom.conf should never be empty when the install finishes.

 

[wardmundy]

Let me preface this by saying we haven't used FreePBX 14 or RASPBX at all.

There is a bug in the del-trunk script if you don't delete the trunks in reverse order, e.g. delete GVSIP3, then GVSIP2, then GVSIP1 not the other way around.

Asterisk 13.22.0 is now part of the installer, replacing 13.21.1. It was released earlier today.

On your platform, there is no /etc/pbx directory so don't worry about the error. You don't have pbxstatus either, presumably.

When the install finishes and you've added a GVSIP trunk, there's no need to reboot. You should run asterisk -rvvvvvvvvvv and then pjsip show registrations to be sure your trunk is registered. If not, check the contents of /etc/asterisk/pjsip_custom.conf to see what came unglued.

The installer builds two types of trunks for every new GVSIP account: PJSIP which is used to make GV calls and CUSTOM which is used in conjunction with creation of Outbound Routes. When you delete a trunk, it doesn't delete the custom trunk which is harmless since custom trunks aren't actually registered anywhere.

FreePBX (other than in Incredible PBX builds) does Module Signature Checking. Your getting errors because we change things. You can turn it off in the Advanced Settings to make the errors go away. Don't do so if you don't have a firewall protecting your PBX.

I'm curious. Does RASPBX build Asterisk from source or from a package?? If it's the latter, it won't work at all with GVSIP since a fresh compile from source is required to add the GVSIP components to PJSIP.

 

[zfpbTM]

Hi Ward,

Thanks for all the hard work! This went pretty smoothly on my box -- until the end. Now I'm getting the error below. I'm running CentOS 7.

Starting Asterisk...
/snip/
[-------------->-------------] 11 secs
[--------------->------------] 12 secs


  [Exception]
  Unable to connect to Asterisk. Did it start?



start [--pre] [--post] [--skipchown] [args1] ... [argsN]



Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
Reloading FreePBX
[B]FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x55e47944e000 < 0x7cf000000000)
FATAL: Make sure to compile with -fPIE and to link with -pie.
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
  1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11[/B]

Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
Reloading FreePBX
[B]FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x559868bcb000 < 0x7cf000000000)
FATAL: Make sure to compile with -fPIE and to link with -pie.
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
  1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11[/B]
GVSIP initialization is complete.

Thanks,

-Matt

 

[randy7376]

@wardmundy

Unless it's changed with a recent release, RASPBX installs Asterisk from packages. My installation is probably about a year old or more. I recently upgraded to Asterisk 13 using the package manager.

/usr/src is empty.

 

[wardmundy]

Hi Ward,

Thanks for all the hard work! This went pretty smoothly on my box -- until the end. Now I'm getting the error below. I'm running CentOS 7.

Starting Asterisk...
/snip/
[-------------->-------------] 11 secs
[--------------->------------] 12 secs


  [Exception]
  Unable to connect to Asterisk. Did it start?



start [--pre] [--post] [--skipchown] [args1] ... [argsN]



Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
Reloading FreePBX
[B]FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x55e47944e000 < 0x7cf000000000)
FATAL: Make sure to compile with -fPIE and to link with -pie.
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
  1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11[/B]

Please wait...

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
Reloading FreePBX
[B]FATAL: ThreadSanitizer can not mmap the shadow memory (something is mapped at 0x559868bcb000 < 0x7cf000000000)
FATAL: Make sure to compile with -fPIE and to link with -pie.
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
  1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11[/B]
GVSIP initialization is complete.

Thanks,

-Matt

I suspect our problems with CentOS 7 have to do with their migration to a new version of MySQL that forces use of InnoDB databases. The current Incredible PBX relies upon the ability to take snapshots of MySQL data by simply copying the files. Unfortunately, that breaks things with InnoDB. Switch to CentOS 6.9 for the time being, and your problems will go away.

 

[wardmundy]

If you used our GVSIP-NAF installer before Saturday, July 14, be advised that you need to add the following new code near the top of pjsip_custom.conf because of reported changes coming to Asterisk 13's PJSIP implementation in a future release. Then issue: amportal restart.

[global]
type=global
debug=true
keep_alive_interval=90

 

[wardmundy]

Incredible PBX 13-13 with GVSIP-NAF for CentOS/SL 6 (beta)

For the pioneers (only), we're ready for you to take a turnkey version of Incredible PBX with GVSIP-NAF for a spin. You'll need a CentOS 6.10 (64-bit) minimal platform to begin. Then just follow the original Incredible PBX 13-13 tutorial substituting the download link below. When the install finishes, you'll have Incredible PBX Lean with built-in GVSIP support but no extensions, routes, trunks, etc. You can run the Enchilada upgrade at this time, if desired, to add 30+ Asterisk apps. It will erase all existing data including any preconfigure trunks, extensions, and routes.

Installation: To get started, login to your CentOS/SL 6 server as root using SSH or Putty. Then...

cd /root
yum -y install wget
wget http://incrediblepbx.com/incrediblepbx-13-13-NAF.tar.gz
tar zxvf incrediblepbx-13-13-NAF.tar.gz
rm -f incrediblepbx-13-13-NAF.tar.gz
./Incredible*

For instructions on obtaining refresh_tokens for your Google Voice trunks, go here. For instructions on creating routes, go here.

Add GVSIP trunks by logging into your server as root and running /root/gvsip-naf/install-gvsip. It's a 10-second install per Google Voice trunk. You'll need your refresh_token and phone number for each GVSIP trunk you wish to add.

Delete existing GVSIP trunks (#1 through #9) by running /root/gvsip-naf/del-trunk. If you need to delete trunks higher than 9, edit del-trunk and add new sections for the number of trunks you have. Always delete trunks in reverse order. We don't recommend deleting GVSIP1, but it works.

Add an Inbound Route for each of your trunks using the 10-digit DID of the GV trunk and specifying a destination for the incoming calls. Except in Enchilada version, inbound calls won't be processed until you add an incoming route.

Add an Outbound Route for each trunk (named GVSIP1 through GVSIPn. Outbound calls will fail until you add an outbound route for the trunk.

To refresh the patched version of Asterisk, copy /etc/asterisk/pjsip_custom.conf to a safe place, delete the contents of pjsip_custom.conf, rerun the installer, and then copy your version of pjsip_custom.conf back to /etc/asterisk and restart Asterisk. That way you won't lose any of your previously configured GVSIP trunks.

 

[dumdedumda]

WM --

Any love with GVSIP for Pogoplug V4 users?

 

[wardmundy]

WM --

Any love with GVSIP for Pogoplug V4 users?

Look through the installer code and sort out the OpenSSL piece for the Pogoplug independently. Once you can get it up to version 1.1 (openssl version), it's probably safe to run the installer AFTER making a good backup. Good luck!

 

[tycho]

WM --

Any love with GVSIP for Pogoplug V4 users?

I was looking into that as well, but haven't yet found a mechanism to install OpenSSL 1.1 on the Debian 7/Wheezy/ARMEL archetecture on the Pogo. Perhaps it is out there, but I haven't yet found it.

As an alternative, I have configured the old-school Python-based ("PYGoogleVoice") "GoogleVoice Callback" on my PogoMobile (actually Seagate Dockstar) and it works like a champ. Let me know if interested and I can send you the scripts. I used what "RonR" had published on DSLReports, trimming it for use with a single GV trunk (more are easy to configure) and tweaking it for use on the Asterisk 11 that is installed from the Nerdvittles Pogo ISO.

 

[restamp]

There seems to be a fair amount of interest in GVsip for the PogoPlugs and/or the ARMel architecture. Personally, I'm using a PogoPlug E2. I did basically what Ward suggests: Since openssl 1.1.0 is not packaged (that I could find), I downloaded the source and compiled it using the configure from the script. I then applied the patches to the latest Asterisk and built that. Everything looked clean (except that there were a lot of patch failures, but virtually all of them were to the READMEs). However, when I run this new Asterisk with the 1.1.0 openssl, I get SSL errors:

[2018-07-15 12:25:06] WARNING[1622] pjproject: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <337092801> <SSL routines-tls_post_process_client_hello-no shared cipher> len: 0

Googling this yields a number of hits, but so far I haven't hit on the magic bullet to make it work.

I'd rather use the NAF approach, but I may give the PY code a try, although I really haven't taken the time to understand it. Right now I'm using OBis for GV access which usually work except for 5-ring
answering machines and voicemails.

If anyone succeeds in getting NAF going on ARMels, please let me know and I'll do the same.