Thanks Rob,
That was my initial thought, the pbx running out of memory.
This pbx is a $5 VM on Vultr 1GB ram. I believe that some users here have the same VM deployed with no issues. I’m not trying to blame it on, but it seems to me that this issue only happens when the pbx firewall is running. Is there any way I can troubleshoot that?
Then I doubt that your jitter buffer is helping you. it is in a fact a buffer that can ‘fill-up’
You are right!
This is happening so randomly, some times only once in a day other days does not happen at all. It has been hard to diagnose and isolate the problem.
I will look into the network adapter and enable linphone’s log. Let’s see…
No. This is not correct. “UCP Node Server” used to be a separate module that was commercially licensed but did not require sysadmin nor did it even require the distro or cent os. In FreePBX 14 UCP node is included with UCP and the separate module doesn’t exist in 14.
The license is also AGLP3. So UCP node is the same license since its part of UCP.
I don’t know what this implies but it’s also wrong.
While I like BladeStudios, and I know he is just misinformed, please take note that he does not work for Sangoma. Any official comment about how any modules work should come from a Sangoma representative. We all have little blue Sangoma symbols in the lower right hand side of our avatars.
The UCP Node Server component, which is part of UCP, and is open source. (see links above). Allows such things as users being able to connect to XMPP over UCP. XMPP is also open source (formerly commercial).
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From the asterisk CLI
rtp set debug ip (suspect host)
you will see any out of order or delayed rtp packets , I suggest you install sngrep and RTFM for it 
CentOS 7 says 2GB min but 4GB recommended last time I checked.
maybe you set weakup call for the phone?
I didn’t catch that they were merged together in v14. So then that does make a difference. Still doesn’t change that NodeJS and the Process Management module is needed for it.
I guess this goes back to the whole “Manual Install” issue I spoke about earlier. The issues in those cases aren’t about the fork of Asterisk or the changes for GVSIP, it’s about not understanding how to fully deal with a manual install and what backend pieces are really missing and need to be installed.
It’s a bit unfair to the OP and those who made this guide to have to answer every little “I followed your guide but…” issue when none of them actually have to do with the part the OP wrote up but just manual install woes in general.
apt-get install nodejs on debian will likely get you
v0.10.29
You might well want at least v8 (you don’t have to eat any veggies though )
A good source I use is:-
https://nodejs.org/en/download/package-manager/#debian-and-ubuntu-based-linux-distributions
which should get you squared away with both node(js) and npm for your js stuff.
No more woe, from that point on you can use npm itself to manage the sub-system.
Have you even glanced at the wiki for installing 14?
^ The section about installing NodeJS. Been there since day one.
Manual install “woes” are the result of people not following instructions. The instructions are on the wiki. The wiki can be edited by outside people if you want it updated to something newer than Debian 8.8.
The “OP” asked about UCP Node. You stated wrong information about it. I merely corrected that information. Not sure why we have to go down the road of how ‘unfair’ FreePBX is… geesh
Same problem with CentOS. So we just use nodesource. 
This is what they were talking about, If you reinstall UCP, then it works until your next reboot, at which point you see this:
node --version
v8.11.3
I am going to tinker to see if I can solve this riddle, anyone have any clues for as to what I should check?
I already tried sudo fwconsole start ucpnode to no avail.
ps aux | grep node
shows nothing running.
personally the options present in the default UCP are all that I need myself, but some users may be looking to download the extra goodies and what not, also if you are unfamiliar and see the red fire icon on your dashboard, that is bound to worry some people that don’t realize UCP will still work regardless.
Yes, I did state wrong information and I was corrected. Not an issue, I’m cool with that.
I never said FreePBX or Sangoma was unfair, I said it was unfair to THE OP who WROTE the guide for using GVSIP has to answer questions about manual installs when people have not read the Wiki guides on said manual installs. I was expressing empathy/sympathy for xekon having to deal with issues that are not related to his guide, that’s all.
then you aren’t starting your system with fwconsole start
https://wiki.freepbx.org/display/HTGS/Example+systemd+startup+script+for+FreePBX
I would suggest to anyone who has followed this thread, that if you want no pain, . . .
A) install FreePBX on any OS of choice , using the WIKI recipes if the ‘distro’ is not feasilble.
B) When it is working, which is hard to dispute, then . . .
C) you can then rebuild asterisk with any ‘patched’ version that works given all the source and tools are in place, it just works that way and has for years.
Putting the cart before the horse has never worked well and always causes these muddleminded disconnects from the obvious route.
JM2CWAE
I saw the Sangoma products from a link on free PBX. Look forward to using them. And thank you for taking the time to Answer. We are new to integrating VoIP but not new to using and implementing it. VoIP is growing at 25 to 28% per year. The Larger VoIP Companys are not a good choice. So it is very important we have a strong stable solution to build around. I will be taking a very close look at Sangoma producs as soon as we are Compatent with FreePBX. I am very happy to hear you have the elastics crew… I used that product once a few years ago for another company. They clearly know there stuff. Thus it is clear we are in great hands with FreePBX and Sagoma.
Jim Bronson
Hi all - I am having some issues around routing, call quality and my vpn configuration. I have a dual-interfaced set up, which is virtualized running within ESXi. One of my interfaces goes directly to the internet, the other goes to an administrative subnet (192.168.1.0/24). I will call this my LAN. I have openvpn set up and running and I’ve also managed to get calls working, but not both at the same time in all configuration scenarios. Here is where it gets interesting. If I have the default route pointed to the WAN’s default route, call quality gets choppy when running a client based on the administrative subnet/LAN (which is connected to another gateway going out to the internet, with its own router). Any clients out on the internet have problems directly connecting to sip, including registrations, etc. OpenVPN connects fine with the 192.168.3.0/24 tun interface on the FreePBX host. It appears then that clients connecting over the VPN are fine (locked out by fail2ban - but that’s a different topic). Call quality, in the short time I could listen before locked out, sounded good.
My other testing surrounded pointing the default route to the 192.168.1.1 administrative subnet/LAN default gateway which is attached to a separate router. In this case, clients on the 192.168.1.0/24 subnet worked well and call quality was vastly improved. But then, OpenVPN stopped being able to connect to clients out on the internet. My assumption is this is happening because the VPN traffic does not know how to return to the internet other than the default gateway already configured with is directed at 192.168.1.1. Is this type of set up not going to work because of the presence of two routable gateways to the internet? Will I be forced in to a situation where I will have to disconnect the admin/LAN side interface to make this work? How are administrative interfaces connected to the LAN in other deployments? I apologize in advance if I have left out any glaring bits of important information. I am happy to share more details. I am eager to solve this problem.
Hi,
I currently have a test freepbx installation for the company I am contracting to. They need an ACD system setup.
One of the requirements is to have some integration with the ticketing system (zendesk). I’ve managed to get AGI scripting to work with Zendesk for calling from IVR etc , but I’m having some issues with the sangomacrm link module.
I have obtained a 1 month free license. Managed to get it installed after some trouble via fwconsole, installing both zulu and sangomacrm edge versions fixed it.
So far so good. However when I switch the crm module settings to REST (there is no option for zendesk , so have to use the generic setting) it then redirects me to another page for the REST settings.
This link however is broken…
We are unable to find any information on the module you are looking for.
url is mysubdomain/admin/config.php?display=sangomacrmrest
Getting this working is going to be a crucial selling point for freepbx , the other option is 3cx , which only has very rudimentary zendesk implementation (and is more expensive).
Any ideas?
Thanks in advance
Hello
I have a new instalation of FreePBX ver 14.0.3.6, (Firmware: 12.7.5-1807-1.sng7).
It is VM on Hyper-v server 2016,
This PBX about 100extensions.
I have a big problem with Cordless Dect Phones GIGASET A510, C610,
If I have more extensions on base station I get status LAGGED (2000-3000ms).
Extensions are random unavailable.
If I use one extension per phone everything work well.
Can you help me someone…
MY extension conf :
[215]
deny=0.0.0.0/0.0.0.0
dtmfmode=rfc2833
canreinvite=no
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
session-timers=accept
icesupport=
encryption=no
namedcallgroup=
namedpickupgroup=
vmexten=
permit=0.0.0.0/0.0.0.0
defaultuser=
rtcp_mux=
dial=SIP/215
secret=trusr645ueytriry
context=from-internal
mailbox=215@device
callerid=215 Hudec Mato <215>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
