@jmmicmc wrote:

I was wondering whether someone could give me some ideas. We have a Freepbx system (framework: 13.0.88.11, Asterisk: 12.12.2), with two POT lines coming in. We have with the following problem: Incoming all lines can be used. If there is an incoming line in use you can still call out. However if someone is already making an outgoing call, the second outgoing call cannot be made. I have done this:

• on the outbound routes I have added the other line as the “Trunk sequence for matched routes”
  We have a Dahdi phone card.

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@Ollie wrote:

Getting the following from cli repeatedly:

Broadcast message from root@xxxx.xxxxx.com (Fri Apr 27 12:47:20 2018):
Loop detected in monitoring script. Dying!

matching entries in /var/log/messages:

Apr 27 12:45:10 freepbx journal: voipfirewalld (Monitor thread): Wall: ‘Loop detected in monitoring script. Dying!’ returned 0

Asterisk: 13.19.1
FreePBX: 14.0.2.18
System Firewall: 13.0.53.3

Happy to provide any other info to help resolve this.

Thanks.

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@JessicaRabbit wrote:

distro version 13-32bit/64bit

After installing 64bit distro and restoring PBX Settings from 32bit installation, incoming calls work but outgoing do not. Outbound routes look ok to me. Need help with log below to track down the problem. Thanks.

2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [2031234567@from-internal:1] Macro(“SIP/300-0000001d”, “user-callerid,LIMIT”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/300-0000001d”, “TOUCH_MONITOR=1524887725.31”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/300-0000001d”, “AMPUSER=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/300-0000001d”, “0?report”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/300-0000001d”, “1?Set(REALCALLERIDNUM=300)”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/300-0000001d”, “AMPUSER=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/300-0000001d”, “0?limit”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/300-0000001d”, “AMPUSERCIDNAME=Operator”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:8] GotoIf(“SIP/300-0000001d”, “0?report”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:9] Set(“SIP/300-0000001d”, “AMPUSERCID=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/300-0000001d”, “__DIAL_OPTIONS=tr”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:11] Set(“SIP/300-0000001d”, “CALLERID(all)=“Operator” <300>”) in new stack
[2018-04-27 23:55:25] WARNING[3031][C-0000000c] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ‘>’, expecting ‘-’ or ‘!’ or ‘(’ or ‘’; Input:
“LIMIT”=“LIMIT” & 3 & 0 & >0 & 0>=
^
[2018-04-27 23:55:25] WARNING[3031][C-0000000c] ast_expr2.fl: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:12] GotoIf(“SIP/300-0000001d”, “0?limit”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:13] ExecIf(“SIP/300-0000001d”, “1?Set(GROUP(concurrency_limit)=300)”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“SIP/300-0000001d”, “0?Set(CHANNEL(language)=)”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:15] NoOp(“SIP/300-0000001d”, “Macro Depth is 1”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:16] GotoIf(“SIP/300-0000001d”, “1?report2:macroerror”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (macro-user-callerid,s,17)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/300-0000001d”, “1?continue”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (macro-user-callerid,s,36)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:36] Set(“SIP/300-0000001d”, “CALLERID(number)=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/300-0000001d”, “CALLERID(name)=Operator”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:38] GotoIf(“SIP/300-0000001d”, “0?cnum”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:39] Set(“SIP/300-0000001d”, “CDR(cnam)=Operator”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/300-0000001d”, “CDR(cnum)=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/300-0000001d”, “CHANNEL(language)=en”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [2031234567@from-internal:2] Set(“SIP/300-0000001d”, “ROUTEUSER=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [2031234567@from-internal:3] Set(“SIP/300-0000001d”, “ROUTEUSER=300”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [2031234567@from-internal:4] GotoIf(“SIP/300-0000001d”, “1?notblind”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (from-internal,2031234567,7)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [2031234567@from-internal:7] GotoIf(“SIP/300-0000001d”, “1?restrictedroute-cfcd208495d565ef66e7dff9f98764da,2031234567,2:outbound-allroutes,2031234567,2”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (restrictedroute-cfcd208495d565ef66e7dff9f98764da,2031234567,2)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Channel ‘SIP/300-0000001d’ sent to invalid extension: context,exten,priority=restrictedroute-cfcd208495d565ef66e7dff9f98764da,2031234567,2
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [i@restrictedroute-cfcd208495d565ef66e7dff9f98764da:1] Goto(“SIP/300-0000001d”, “bad-number,s,1”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (bad-number,s,1)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [s@bad-number:1] Goto(“SIP/300-0000001d”, “11,1”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx_builtins.c: Goto (bad-number,11,1)
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [11@bad-number:1] ResetCDR(“SIP/300-0000001d”, “”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [11@bad-number:2] NoCDR(“SIP/300-0000001d”, “”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [11@bad-number:3] Progress(“SIP/300-0000001d”, “”) in new stack
[2018-04-27 23:55:25] VERBOSE[3031][C-0000000c] pbx.c: Executing [11@bad-number:4] Wait(“SIP/300-0000001d”, “1”) in new stack
[2018-04-27 23:55:26] VERBOSE[3031][C-0000000c] pbx.c: Executing [11@bad-number:5] Playback(“SIP/300-0000001d”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2018-04-27 23:55:26] VERBOSE[3031][C-0000000c] file.c: <SIP/300-0000001d> Playing ‘silence/1.ulaw’ (language ‘en’)
[2018-04-27 23:55:27] VERBOSE[3031][C-0000000c] file.c: <SIP/300-0000001d> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2018-04-27 23:55:30] VERBOSE[15571] chan_sip.c: Extension Changed 300[ext-local] new state Idle for Notify User 302
[2018-04-27 23:55:30] VERBOSE[3031][C-0000000c] file.c: <SIP/300-0000001d> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)

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@vesperbot wrote:

We have decided to try employing FreePBX’s module “XMPP server” for intra-org instant messaging, and have found out that it does not listen on port 5222/tcp, but listens on 5222/tcp6. I haven’t found a config entry to make it listen on ipv4 in /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat/settings.yml (it only has “port” and “domain” entries passed to node-xmpp-server), so I went mad and tried to disable IPv6 everywhere - wow, no wai, netstat -nlp still shows letschat listening on “::” instead of either 0.0.0.0 or an interface. How to make this piece of software obey OS restrictions and either first try creating a socket on ipv4, or just forget about ipv6?

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@steve_pbuk wrote:

I set a FreePBX server up with 20gb Hard Drive (mainly for testing) and now I want to add a 2nd hard drive and use this for call recordings.

Do you recommend adding the hard drive and reconfiguring FreePBX to use it for call recordings or would you recommend just mounting the new drive to the default call recording folder?

I’m using the latest distro version.

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@adminpbx wrote:

Dear all my issue is very simple but I am unable to fix. I have a callcentric account and our offices have USA/UK/AUS numbers for customers i.e. people in USA call on our USA number, people in UK on our UK number etc. and all these calls terminate via callcentric to our office.
Previously within callcentric we had few DIDs setup so that if call comes from
US number it would go to ext 10, 11, 12
UK would go to ext 20, 21, 22
AUS would go to ext 30, 31, 32
or in simple words, routing based upon Called ID (my USA, UK, AUS number for customers)

Now, as you know cc gives your one number 1777XXXXXXX
So how I setup freepbx for same as I can put this 1777XXXXX number in one trunk.
I know some other people have also raised this issue but one problem that I am facing is that I noticed is that the caller ID would always be my test mobile number and hence I could not route calls based upon caller ID (which is what I have to achieve). My freepbx is unable to forward the callcentric DID, it’s always forwarding my test mobile. And I can forward DID of callcentric only for one.

I have freepbx latest version with asterisk 13

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@JessicaRabbit wrote:

What worked for me:

1. Created full disk image backup of 32bit v13
2. Created final FPBX full backup to FTP server (remember the FTP config).
3. Installed 64bit v13 Distro, brought up to date and restored the FPBX setting from FTP backup.
4. Reviewed and confirmed the operation of the 64bit version.
5. Created full disk image backup of 64bit v13.
6. Created final FPBX full backup to FTP server (remember the FTP config).
7. Ran the 13 to 14 tool following the instructions for SSH to root and commands.
8. Watched with great alarm all the warnings and errors on the console hoping they were not a scary as they appeared.
9. Apparently they worn’t as system booted up and seems normal so far, amazing.

I am now running v14.

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@wesleyg05 wrote:

Freepbx newbie. I am dropping inbound calls from Broadvoice. All outbound calls work fine. I have looked over everything and can not seem to find an issue.

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@VoipTom wrote:

Hello,

I can’t register the sip trunks (chan_sip or chan_pjsip). I get the error:
NOTICE[2152] chan_sip.c: – Registration for ‘0xxxxxxxxx@tel.t-online.de’ timed out, trying again (Attempt #85)

I use a Laptop behind a speedport-nat-hybrid router. For testing I use a softphone (phonerlite) on the same Laptop. This voip-client is working fine.

If phonerlite can use the trunks, why can’t freepbx this ?

Thank you !!
Tom

FreePBX Error.jpg1814x1087 273 KB

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@smsfabio wrote:

Hello,

I’m setting up the IVR.

however I would like to add the option that, when the client does not type anything, it even then routes to an extension.

What do I have to type in Digits Presset to have this function?

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@MacsOffice wrote:

I have the IVR all set with options 1 - 8. The extensions are all 2xx (3 digits) . All seems fine to me.
I cant seem to get the delay right to reproduce this, but the complaint I hear is that “every” time someone direct dials an extension - choice #2 of the IVR is selected instead.
Its obviously grabbing the first digit and running with it. I recall there is a “wait for more key presses” kind of setting somewhere, is that what I am looking for and where is it?

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@RKFAULHABER wrote:

Hi,
One of our remote VOIP phones is suddenly getting multiple calls from extensions that do not exist.
On the user’s phone it will list extensions like “100”, “101”, “1000”, “200”, etc… On some, it
showed the extension as “‘hi’ OR ‘x’=‘x’”.
These calls do not show up in the CDR log.
I looked in our FreePBX settings and see that under Settings – Asterisk SIP Settings, Allow Anonymous SIP calls is set to NO.

I suspect someone is trying to hack into his phone on his local network.
Does anyone have insight into this?

Thank you!
-RKF

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@leegreen01 wrote:

I am trying to register a google voice account via oauth 2 and it always shows as disconnected if i use plain text it works fine.

The error i see is below

[2018-05-02 18:56:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 18:57:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 18:58:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 18:59:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:00:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:01:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:02:17] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:03:18] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:04:18] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”
}
[2018-05-02 19:05:18] ERROR[10064] res_xmpp.c: An error occurred while performing OAuth 2.0 authentication for client ‘gandyall-tradecom’: {
“error”: “unauthorized_client”,
“error_description”: “Unauthorized”

I have double checked all the information from google seems to be entered correctly but as i cannot find any examples i have nothing to compare against.

Is there a guide or any further information on how to get this working ?

any help would be much appriciated

Thank you

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@davidbqzt wrote:

Hi,

I have to install FreePBX on some kind of VPS or “CloudServer”, that server has CentOS 6.9, I have noted some particularities, but I can’t do nothing about it, is the server that have been provided, so I have followed the installation guide on https://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+6, first on a virtual box with CentOS 6.9 minimal ISO installed and I succeded to install every thing, then I try it on the VPS, but I got stuck on ‘make config’:

contrib/scripts/install_prereq install
this command fail to install uw-imap-devel.and hoard, both on VPS or VirtualBox, I’ve try to find a lot of packages but, finally I omit them
./configure --libdir=/usr/lib64
Succeded
contrib/scripts/get_mp3_source.sh
Succeded on VirtualBox fail on VPS with error: svn: PROPFIND of ‘/svn/thirdparty/mp3/trunk’: Could not read status line: Connection reset by peer (http://svn.digium.com)
make menuselect
Both Succeded
make
Both Succeded
make install
Both Succeded
make config
Succeded on VirtualBox failded on VPS with error: /bin/sh: /etc/os-release: No such file or directory
ldconfig
Succeded on VirtualBox, not tried on VPS

So I have two errors, I can live without MP3 but not without make config.

It seems like if the /etc/redhat-release wasn’t present, but it is and is correct.

Anybody have any idea of what could be or what could be done?

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@chemnic wrote:

Probably the title is wrong, but here I try to explain my problem:

I want my users to access UCP with WebRTC enabled and working. I know that for that to happen I need a certificate which has to be accepted by the browsers.

What I do for my other APPS (nextcloud, odoo, etc.) is to use a reverse proxy server which also generates Let’s Encrypt certificates. I have just one public IP address and the port 80 in this IP is forwarded to the reverse proxy server.

What I did, for simplicity, was to access the PBX directly without reverse proxy. I manually copied the certificates (eg. pbx.mydomain.com) from the reverse proxy to the PBX server, then applied the certificate to UCP and apache from the GUI and everything is working great.

UCP is working fine, with the WebRTC and all.

The problem is that this certificate is going to expire soon, what is normal in Let’s Encrypt, but I won’t have the benefit of auto renewal as I have in my other apps because I migrated the certificate manually. I will have to do it again every time the certificate is near expiration.

I want your suggestions of how can I automate this process. I can have a script to copy the certificates daily, but I don’t know exactly where to put them or if I need some extra work. Also I think I should put the certificates in apache configuration. Thank you.

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@snaplink wrote:

Current PBX Version 14.0.2.14
PBX Firmware: 12.7.4-1804-1.sng7
PBX Service Pack: 1.0.0.0

I have a successful implementation of FreePBX up and running for a couple weeks now and one thing I keep getting asked about is if there is a way to search for an extension on the handset itself. We’re using Yealink T46G phones. So far, from what I can tell is that if the extension isn’t a BLF or speed dial on the phone already you can’t search the extension list that is in FreePBX.

Is that correct or did I miss the boat completely?

Thanks!

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@PCS wrote:

Hi all,

We’ve had several occurrences of hour-plus-long voicemails being left for users on various systems. The most recent was on a FreePBX 12.0.76.2.

In the Voicemail admin, there were no settings for maxmessage or maxsecs, either in the global or user-level. My understanding is the default is 5 minutes (300s).

I checked the CDR logs, where I could listen to the messages, and all of the offending callers were fax machines.

Why would asterisk be ignoring the max message length? I’ve manually set them to 300s to see if it makes a difference, but since this has happened on multiple installations, I’m a tad concerned.

Also, I can’t find a way to delete those recordings from the CDR. I can’t find them by the filename on the filesystem, which makes me believe they’re living in the database. Is that the case?

Thanks,

Brad

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@damania wrote:

All I’m trying to do is get sms/IM/txt/chat to work on Bria or Zoiper softphones but I have been unsuccessful in putting together an elegant solution.

Yes, XMPP (by adding two accounts into the phone, one for SIP and one for XMPP) works but on a different extensions! I need calls and txt to work on a single extension. Why was this overlooked in development? If UCP is getting in the way of me creating an user that’s the same as an extension how do I fix it? I don’t need UCP login features!

Maybe simpler is to get SIP SIMPLE to work? How would I do that? Is there a module?

Lastly, if FreePBX can’t give me what I need, what other distro or pbx can I implement paid or unpaid that has a slick integrated android/ios app?

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@sp9 wrote:

I’m in the process of ‘playing’ and learning about FreePBX. I’m new to phone systems in general. So far, I’ve got FreePBX installed and I have 2 Cisco SPA 504G phones to use. I went ahead and purchased CM Endpoint Manager and I’ve got a phone template set up for the SPA504Gs, local networks configured/trusted, extensions created, phones assigned, etc. and the phones autoprovision via TFTP and they work! I can call each extension, use voicemail, DTMF is working with a test IVR, I have an outbound route set up with a prefix that isn’t used with any trunks yet, etc. I’ve only used PJSIP so far w/ port 5160.

Here is my network layout for FreePBX
-172.16.1.1/24 pfSense LAN1
-172.16.1.4 FreePBX interface (only interface on server)
-172.16.200.1/24 pfSense LAN2
-172.16.200.101-254 DHCP range on LAN2 for testing phones, with option 66 to point to FreePBX TFTP

• +WAN gateway on pfSense, ex. 72.81.4.253 public IP (not mine, just picked random for example)
  **To keep it simple, all LAN to LAN traffic is currently allowed…

I’d like to test out connecting a few phones into the FreePBX from outside of the local network. If I use FreePBX for a production solution, I’ll have to accommodate 10 remote SOHO users. For site-to-site, I would simply use VPN tunnels, and allow them to communicate through the LAN to FreePBX (like trusted).

I’m not quite sure how the phones connect to the PBX yet, but I thought that the details must be in the TFTP config- which comes from the phone template/mapping? The details being username/secret and SIP gateway?

I will need to figure out how to manually update these “soho remote” phones with the WAN gateway, find out what will need updated in the phones, and then how to configure FreePBX side to allow connections from WAN, only certain extensions/users or templates if possible, set up port forwards UDP 5160, RTP 10K-20K from WAN IP to FreePBX LAN IP, …should I use TLS/SRTP, etc… so many things. I think one option for configuring phones would be through their web interface they’ve got…

My question is, even if I can get “soho remote” phones working and get a good understanding of how and why it works- is it safe? I’m not sure how I would connect these remote users otherwise, unless I sent them all VPN boxes to put behind their routers, or purchase phones capable of tunneling or OpenVPN…

-How to configure phones for remote users without TFTP?
-What will need configured on phones?
-What will need configured on FreePBX to allow them after setting up port forwards from router WAN IP?
-Is it safe for them to hit WAN IP without VPN relying on port forwarding?

I know this is sloppy, but I’d appreciate any guidance/advice. Thank you…

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@Gsmitt wrote:

Hello all,
I’m have troubles with my User control panel.
Error” Unable to connect to the UCP Node Server because: ‘Error: xhr poll error’

001-freepbx-ucp.png1368x487 15.9 KB

My environment as follow;
CentOS 7 Virtual machine, 2 CPU, 4GB Ram, 100GB HDD
ISO Used= SNG7-FPBX-64bit-1712-2.iso
FreePBX 14.0.3.1
Asterisk 13.0.7.1
User Control Panel14.0.2.5

I installed FreePBX on Virtual machine, Logged in Admin console then executed all updates for modules and the system. Once updates were finished, I created EXT (101), enable UCP. With http I had no problems. I wanted to test Softphone system on UCP, it requests HTTPS, I configured self-signed cert as show in image URL above. After logging back in I received the error “Unable to connect to the UCP Node Server because: ‘Error: xhr poll error’”

Research the following places;

https://issues.freepbx.org/browse/FREEPBX-11186


The following steps have been executed;

1. fwconsole stop ucpnode
2. fwconsole start ucpnode
3. yum remove nodejs
4. yum install nodejs
5. I tried fwconsole ma uninstall/install/download ucpnode
   
   002-fwconsole-uninstall.png864x553 9.77 KB
6. Restarted the server after making Certs, completed
7. Uninstalled XMPP
8. Installed XMPP again
   Dashboard
   
   003-dashboard.png1782x663 79.7 KB
   
   Port Management
   
   004-port managment.png1853x745 78.1 KB
   
   Any more adice or direction would be appreciated.
   Thanks in advance

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