Well that is unexpected. Update to edge with:
fwconsole ma upgrade sangomacrm --edge
fwconsole r
If problem persists take two asprin open a commercial module ticket:
https://support.sangoma.com/
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Problems with sangomacrm REST settings
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Source IP address not in ACL
possibly its in /etc/hosts, possibly your DNS server is not truly functional, Use the raw IP instead.
but, significantly , from my machines
ping cclab.ptsn.us1.twilio.com
ping: unknown host cclab.ptsn.us1.twilio.com
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Ringing is "static-y" on incoming calls and dialing on keypad
Hi Jessy:
There is a ticket now on what I would call cosmetic audio issues with ring tones and DTMF tones. It should be resolved in a firmware update in the near term.
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Ringing is "static-y" on incoming calls and dialing on keypad
YES! That is exactly what it is. the DTMF tones and ring tones. 
So i am not crazy!!! Thank you so much Lorne appreciate it.
I will let my clients know.
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Source IP address not in ACL
We did an fwconsole restart and that seems to have taken care of this particular issue. Not sure why that would make any difference if we applied the changes. Thanks for the suggestions.
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Can only receive inbound after making an outbound call
This is almost always an indication that the NAT session is expiring. Once you make an outgoing call, a new session is established, which allows incoming calls once again, that again fail after the session is expired and the cycle repeats.
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Source IP address not in ACL
Consider it a birthday present.
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Recommendation: Soft phone for use by Operator/receptionist?
I have tried Zulu before… really didn’t like it 
That was a while ago, though, so I may take another look.
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Follow-Me: Calls not going to cell phone VM
OK, so it looks like I have this resolved, but you are not going to believe what the solution is. The DID I was testing with that worked as expected has “Fax Detection” enabled, and the DID of the user having the problems did not. I enabled “Fax Detection” on the user’s DID, applied the config and tested. FM started working as expected.
WTF?
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
FWIW…
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ZERO sample files are required. After installing Asterisk (NAF’s or otherwise), FreePBX does require at least an empty modules.conf (and possible an empty ari.conf) or it gets upset. NAF’s Asterisk requires an empty statsd.conf, but that’s the extenit of it.
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I have no problems with UCP after installing FreePBX 14 and Asterisk straight from NAF’s repository:
And it comes up reliably after a reboot:
root@FreePBX:~# ps aux | grep node
asterisk 1575 1.8 4.5 126312 45172 ? Ssl 20:22 0:02 node /var/www/html/admin/modules/ucp/node/index.js
When I install nodejs, I also install libicu-dev (and use nodejs v9, but v8 works equally well):
curl -sL https://deb.nodesource.com/setup_9.x | sudo -E bash -
apt-get -y install nodejs
apt-get -y install libicu-dev
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
Thank you I appreciate this info. I did find out that modules.conf was required, I figured it out the hard way, removing a few .conf files at a time until I figured out which ones I needed to make it run. Going to look over what you’ve posted and try some stuff out.
EDIT: I included the sample file smdi.conf because without it, FreePBX ran, seemingly without issue, but voicemails did not work, I tried loading the voicemail.so file manually as per an example and found that it complained about smdi (a dependency) not yet being loaded, after copying the smdi.conf file from the samples and issuing a reboot, the voicemails started to function again.
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
what does your ps aux | grep asterisk look like?
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
I don’t have an smdi.conf on any system and voicemail works fine. If you’ve installed SMDI (Simplified Message Desk Interface), then I suppose you may need smdi.conf.
root@FreePBX:~# ps aux | grep asterisk
asterisk 750 1.1 3.6 158056 36504 ? S 20:22 0:50 /usr/sbin/apache2 -k start
asterisk 751 1.1 3.6 158848 36660 ? S 20:22 0:50 /usr/sbin/apache2 -k start
asterisk 753 1.0 3.3 157624 33392 ? S 20:22 0:47 /usr/sbin/apache2 -k start
asterisk 1339 0.1 3.3 117644 33052 ? Ssl 20:22 0:07 PM2 v2.10.1: God Daemon (/home/asterisk/.pm2)
asterisk 1475 1.1 3.6 158448 35956 ? S 20:22 0:50 /usr/sbin/apache2 -k start
asterisk 1476 1.0 3.7 158060 37756 ? S 20:22 0:47 /usr/sbin/apache2 -k start
asterisk 2292 1.1 2.8 157128 28772 ? S 20:23 0:49 /usr/sbin/apache2 -k start
asterisk 3782 1.0 3.3 157980 33676 ? S 20:26 0:46 /usr/sbin/apache2 -k start
asterisk 10644 0.9 3.6 158996 36008 ? S 20:40 0:31 /usr/sbin/apache2 -k start
asterisk 23840 0.0 2.5 157192 25720 ? S 21:10 0:00 /usr/sbin/apache2 -k start
asterisk 23971 0.1 1.3 45424 13176 ? Ss 21:10 0:01 dirmngr --daemon --homedir /home/asterisk/.gnupg
asterisk 24277 0.0 2.7 157456 27028 ? S 21:11 0:00 /usr/sbin/apache2 -k start
root 24857 0.0 0.0 1900 112 pts/0 S 21:13 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk 24859 1.1 5.5 166252 55216 pts/0 Sl 21:13 0:16 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
asterisk 25019 0.3 4.0 127348 40240 ? Ssl 21:13 0:05 node /var/www/html/admin/modules/ucp/node/index.js
asterisk 29220 0.0 0.0 1900 400 ? Ss 21:38 0:00 /bin/sh -c [ -x /var/www/html/admin/modules/dashboard/scheduler.php ] && /var/www/html/admin/modules/dashboard/scheduler.php > /dev/null 2>&1
asterisk 29223 8.7 2.7 90116 27652 ? S 21:38 0:00 php /var/www/html/admin/modules/dashboard/scheduler.php
root 29235 0.0 0.0 4372 552 pts/0 S+ 21:38 0:00 grep asterisk
root@FreePBX:~# ps cax | grep asterisk
24857 pts/0 S 0:00 safe_asterisk
24859 pts/0 Sl 0:16 asterisk
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
cool thanks, interested to see the difference between an install done using root vs not, yours shows
root 24857 0.0 0.0 1900 112 pts/0 S 21:13 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk 24859 1.1 5.5 166252 55216 pts/0 Sl 21:13 0:16 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
while mine shows
asterisk 731 0.5 4.2 2046712 84696 ? Ssl 18:44 0:16 /usr/sbin/asterisk -U asterisk -G asterisk
I notice your safe process is run as root, but then you have another ran as asterisk, I think that is what they refer to as the process forking.
I also see that yours has these parameters that mine does not: -f -vvvg -c
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
I suspect you didn’t install a service (in init.d) that starts Asterisk using ‘fwconsole start’ and stops Asterisk using ‘fwconsole stop’.
usr/sbin/safe_asterisk should always be running and is used to automatically relaunch Asterisk if it crashes for some reason.
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
FWIW, I have NAF’s GVSIP implemented on the Raspberry Pi and have no problems sending and receiving faxes at 14400 w/ECM through Google Voice using HylaFAX. I’ve never tried the FreePBX fax module.
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Misc destinations not working
Something to consider, thanks for the heads up!
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How-To Guide for Google Voice with Freepbx 14 & asterisk gvsip, Ubuntu 18.04
Confirmed, I just did the install with zero sample files as well.
So I just installed this as Per the wiki (the freepbx install part with “sudo su”), and it seems that having an empty modules.conf file is the solution that I have been looking for from the start of all this samples file confusion.
Unfortunately nobody ever suggested trying this. I got to the point that I narrowed down the required sample files for my install method down to a couple files, and at that point you mentioned the empty modules.conf file needed (which is better than using the sample one) Also earlier in the thread you pointed out the required empty statsd.conf file.
I just tried this myself, and the entire issue when I was trying to install this following the wiki was because of the missing modules.conf file. A lot of time could have been saved had I realized this sooner.
I did learn some stuff in the process of working on this, so the time spent was not a complete waste. I also devised a very nice backup and restore solution, in which I can boot to another copy of ubuntu, issue a backup of a partition in about 30 seconds (or restore the partition in about 60 seconds) and then boot back to my main copy of ubuntu, all VIA SSH, without even touching the remote computer or any usb thumb drives.
I have updated the existing wiki and modified the touch line to include the other empty .conf files.
going to clean up this guide one more time.
Also, I appreciate you posting @reraikes it is helpful information 
After rereading my post, it seemed I was frustrated, and I just want to make sure you know I am not frustrated with you It is just frustrating when the reason for something not working is not at all obvious.
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PBXact Asterisk Phonebook
Thanks Lorne, this worked great.
This is a pbxact appliance and it has the phone book module installed. Do versions matter? It’s about a year old.
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Hot Spare Firewall Option
Hello All,
I know in the hot spare server you restore here on the spare system, however there is no option to ignore firewall polices on restore. Everyone’s situation is different, but I see times where we don’t want to restore the primary firewall because the environment is different.
This also means that if you want to accept something from a certain IP range on the Hot Spare, you will have to program that same firewall policy on the primary, so it makes it to the spare.
@xrobau I know this was your brainchild at one point, not sure if you have any workaround or suggestions, or if this makes sense to when there is time add it in to the restore here process as an option to exclude.
Thanks All!
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