ALERT GV: The Sky Has Fallen... Really

Ward (and anyone else with an opinion): Given the choice between 13-13 on CentOS 6 and 13-13 on Ubuntu 18.04, which would you choose and why?

Ubuntu 18.04 is very new code, and there were some hacks to get PHP 5 and ODBC working properly with Asterisk and FreePBX. I'd stick with CentOS 6 if it were my call simply because it has a proven track record and is rock-solid reliable.
 
@SMTC: A lot has changed since that release. I'd start over just to be sure I had a reliable platform. Because of all the changes, I think cut-and-paste is your safest bet for moving things over.
 
Following this tutorial for another server, install-gvsip did not add gv trunk to pbx server and not obihai device in google account.
 
Ubuntu 18.04 is very new code, and there were some hacks to get PHP 5 and ODBC working properly with Asterisk and FreePBX. I'd stick with CentOS 6 if it were my call simply because it has a proven track record and is rock-solid reliable.

This past week I attempted two installs on CentOS (deets below). Both failed amusingly, so I am now trying it a third time on Ubuntu. Surely my error, or the fault of my VPS provider (our benchmark of excellence: CloudAtCost), but given those failures I'm trying Ubuntu 18.04 as I type this.

***

Deets: I twice this week attempted to create a "newest-of-the-new" I-PBX installation on (yeah, yeah, but I have the resources lying around, they are working fine at the moment, its a sandbox to play in and I don't think it is the source of this problem) Cloud at Cost. I've done this many times over the years without issue. I used to always use CentOS. Then a few years back, because of difficulties getting a CentOS server spun-up on CaC I moved to Ubuntu. This time, because of difficulties upgrading the CaC-template Ubuntu 14.04 image to 18.04 (/boot is too small and runs out of room; have to fiddle with resizing partitions which I didn't initially want to do but now have done), I reverted to trying CentOS.

I created a base CentoOS server. It is installed as ver. 6.7. The first time I brought it up to 6.10 myself. The second time I let the I-PBX script ("yum -y update") make that happen. Fine and good: CentOS 6.10 was up and running.
I invoked the remaining parts of the script [see http://nerdvittles.com/?p=23948 > Installing a Base CentOS Operating System > Installing Incredible PBX 13-13 LEAN > "Once you have CentOS up and running..."]. I ran the script [./IncrediblePBX-13-13.sh] and then ran it again. Automatic Update Utility ran. Status menu displayed, but showed Asterisk as down. I ran "/root/admin-pw-change" and "/root/timezone-setup."

Couldn't access the Incredible PBX 13-13 Web GUI -- nothing responded (500 error). Couldn't start/stop Amportal/Asterisk. 'Status' could not find the asterisk installation even though I thought I saw it properly installed in the flurry of compilation (I did notice the ascii-art "@" come up during the scroll).

So, the error happened exactly the same way, twice. I was going to seriously trouble-shoot the errors, but then I decided to just bite the bullet, blow-up that attempt (thus losing the logs), create a new Ubuntu server and re-partition the default CaC Dev1 template to give me the room that I needed to upgrade it to 18.04 (ultimately not terribly difficult. Shrank the swap partition by 200 MB and donated that to the /boot partition).
 
I did a brand new installation of a incrediblepbx13-raspbian8-gvsip.zip on a Raspi 3. I followed the tutorial on install-gvsip, everything went smooth. I can see the obihai device in my google voice account. I can call me on the number. I have only one inbound route and it is configured to take ANY call of ANY trunk. So that works. BUT I can't see the GVSIP Trunk at Connectivity > Trunks and therefore, if I want to define a outbound route, using the GVSIP, I can't as it doesn't show up under Trunks and so I can't choose it to configure an outbound route. What to do???
 
Ubuntu 18.04 install went smoothly.

Spin-up CloudatCost VPS server (1 vCPU/512M RAM/10G disk -- installs with Ubuntu 14.04 OS and a /boot partition that is only about 140M). Screw around with deleting preconfigured 1G swap partition to make room for extending /boot partition to 350M so that there is room for OS upgrade (tried; can't be done with a weenie /boot partition), then rebuild smaller swap partition. Upgrade from Ubuntu 14.04 to 16.04, and then from 16.04 to 18.04 (can't go directly from 14.04 to 18.04). Install latest IncrediblePBX lean. Get GVSIP token. Install GVSIP. Configure extension and outbound route. Don't want inbound calls to my GV number coming into the PBX -- want to use it only for outbound and have inbound continue to forward to CC Free NY DID > home ATA. Notice that there is nothing relating to GVSIP on the GV portal. Guess, correctly, that I'd see it on the Legacy portal. Yup; it's there; turn GVSIP-Obi reference off as a forward-to phone. Configure 3CX desktop SIP (not PJSIP) client with PBX credentials. Make a couple of calls. All is well...
 
This past week I attempted two installs on CentOS (deets below). Both failed amusingly, so I am now trying it a third time on Ubuntu. Surely my error, or the fault of my VPS provider (our benchmark of excellence: CloudAtCost), but given those failures I'm trying Ubuntu 18.04 as I type this.

Couldn't access the Incredible PBX 13-13 Web GUI -- nothing responded (500 error). Couldn't start/stop Amportal/Asterisk. 'Status' could not find the asterisk installation even though I thought I saw it properly installed in the flurry of compilation (I did notice the ascii-art "@" come up during the scroll).

Once you finish the install, you MUST reboot your server before Asterisk will come to life. For some reason, it gets hung during the install and cannot be restarted with amportal restart because the half-running version cannot be cleanly killed.
 
Does anyone have an idea on the problem I described on my above posting???

FreePBX knows nothing about GVSIP trunks. It can't create them and doesn't even know when they are running. You have to create them with /root/gvsip-naf/install-gvsip utility. Part of that setup process also creates a matching Custom Trunk (gvsip1-gvsipn) to use with FreePBX. Once you complete the trunk setup procedure, you then can go into FreePBX and add an Inbound Route and Outbound Route. Complete tutorial here.
 
Following this tutorial for another server, install-gvsip did not add gv trunk to pbx server and not obihai device in google account.

Not enough information to be of much help. Check the install log in /root for hints.
 
Once you finish the install, you MUST reboot your server before Asterisk will come to life. For some reason, it gets hung during the install and cannot be restarted with amportal restart because the half-running version cannot be cleanly killed.

Thanks; I don't recall reading the underlying explanation before. I'm pretty sure you stress the point of rebooting in the tutorial, and I'm pretty sure I did reboot -- I've followed your tutorials with excellent results dozens of times over the years -- but who knows; I am getting old. If I neglected to reboot twice in a row it could be a sign of impending issues on my part. :)

All working swimmingly because for the third attempt I went with Ubuntu, which I've been using in lieu of CentOS/SL for perhaps 4 years.
 
That is what I did and it worked for me. I just ran the script. I removed my old GV Motif trunks before I started just to be safe. Also, just out of caution, I backed up my FreePBX configs, recordings and such as my previous backups were out-of-date. However, it wasn't necessary as the upgrade was flawless. One caveat I found out is I had long ago disable PJSIP and you obviously need that to make GV work with the new PJSIP method.

Will this "install-gvsip" script work, if you just click "Disable Trunk (Check this to disable this trunk in all routes where it is used)" for each of the trunks which the "Connectivity > Google Voice (Motif)" module has auto generated, ("GVM_tendigitphonenumber") ?
 
Not enough information to be of much help. Check the install log in /root for hints.
Hello
Today I install ubuntu with gvsip support. Incoming and outgoing call good. I examine the log file, there are 37170 line.
1. Web min not install, I receive 55 warning.
2. I was installed ubuntu with Disk sda5_crypt. Put to password during boot time. How do i disable this or I have to reinstall.
 
Also:
I read the code in the GVSIP branch on NAF419 (Nick French)'s github.
GVSIP only modifies the PJSIP / PJPROJECT code.
So GVSIP should work on all asterisk versions that support PJSIP.
This means this GVSIP (PJSIP patch) should work fine on Asterisk 13, 14, 15.
Has anyone tried installing and running this GVSIP patch on Asterisk 15 ??
Results??
 
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Will this "install-gvsip" script work, if you just click "Disable Trunk (Check this to disable this trunk in all routes where it is used)" for each of the trunks which the "Connectivity > Google Voice (Motif)" module has auto generated, ("GVM_tendigitphonenumber") ?

I imagine it would. I deleted mine but like you said the GV SIP uses PJSIP and the old GV trunks used Motif/XMPP
 
I imagine it would. I deleted mine but like you said the GV SIP uses PJSIP and the old GV trunks used Motif/XMPP
Thanks @Jake
@wardmundy what's your opinion, if you just disable the obsolete GVmotif/xmpp trunks in the ipbx web portal as I describe above, would the install-gvsip script work ok, in other words, not configure the ipbx into an impossibly broken state?
 
Other than possible confusion to the user, there is no reason to disable or delete the motif trunks & accounts. GVSIP and Motif have zero dependency or conflict between each other.

On the other hand the Motif accounts/trunks are useless at this point, so I don't know of any valid reason keep them.

I guess you could keep the Motif custom trunk definitions and change the dial stings to point to the appropriate gvsip instances, then you shouldn't have to touch any in/out routes. We elected to delete them to avoid confusion, I wanted it immediately clear a system was no longer on Motif. I guess we could have just edited the dial string renamed them.
 

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