@Bifur wrote:

Hey everyone,

So an SSH backup was performed today to our backup PBX. Once it completed we lost access to the web gui (on backup server) and it displays an HTTP 500 error. I've searched for similar issues but none have successfully resolved the issue.

I have tried amportal chown but that did not make a different.

If I try to upgrade modules from the command line I get the following:

PHP Fatal error: Uncaught exception 'Exception' with message 'Unable to locate the FreePBX BMO Class 'Userman'A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) amportal a ma install userman 2) amportal a ma enable userman' in /var/www/html/admin/libraries/BMO/Self_Helper.class.php:205
Stack trace:

0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(100): Self_Helper->loadObject('Userman')

1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(36): Self_Helper->autoLoad('Userman')

2 /var/www/html/admin/modules/userman/functions.inc.php(6): Self_Helper->__get('Userman')

3 /var/www/html/admin/modules/restapi/functions.inc/userman_hooks.php(20): setup_userman()

4 /var/www/html/admin/modules/restapi/functions.inc.php(15): require_once('/var/www/html/a...')

5 /var/www/html/admin/bootstrap.php(212): require_once('/var/www/html/a...')

6 /etc/freepbx.conf(9): require_once('/var/www/html/a...')

7 /var/lib/asterisk/bin/module_admin(15): include_once('/etc/freepbx.co...')

8 {ma in /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 205

In the HTTP logs I see it as well:

[Thu Sep 07 14:42:03 2017] [error] [client 24.96.209.3] PHP Fatal error: Uncaught exception 'Exception' with message 'Unable to locate the FreePBX BMO Class 'Userman'A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) amportal a ma install userman 2) amportal a ma enable userman' in /var/www/html/admin/libraries/BMO/Self_Helper.class.php:205\nStack trace:\n#0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(100): Self_Helper->loadObject('Userman')\n#1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(36): Self_Helper->autoLoad('Userman')\n#2 /var/www/html/admin/modules/userman/functions.inc.php(6): Self_Helper->_get('Userman')\n#3 /var/www/html/admin/modules/restapi/functions.inc/usermanhooks.php(20): setup_userman()\n#4 /var/www/html/admin/modules/restapi/functions.inc.php(15): require_once('/var/www/html/a...')\n#5 /var/www/html/admin/bootstrap.php(212): require_once('/var/www/html/a...')\n#6 /etc/freepbx.conf(9): require_once('/var/www/html/a...')\n#7 /var/www/html/admin/config.php(99): include_once('/etc/freepbx.co...')\n#8 {main}\n in /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 205

Any thoughts on how I can get back into the GUI? Both servers are the same (hardware and software) and they are the FreePBX12 distro 12.0.76.4

Thanks for any help!

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@grantpasley wrote:

Hi,

I am running FreePBX 14.0.1.4. In previous versions I have configured Openvpn to start with "service openvpn on" and "service openvpn start"

Trying this now in the latest version based on Centos 7 I run :"systemctl enable openvpn" and "systemctl start openvpn" but in both instances I get an error like:

[root@pbx /]# systemctl start openvpn
Failed to start openvpn.service: Unit not found.
[root@pbx /]# systemctl enable openvpn
Failed to execute operation: No such file or directory
[root@pbx /]#
[root@pbx /]# systemctl list-unit-files | grep openvpn
openvpn-client@.service disabled
openvpn-server@.service disabled
openvpn@.service disabled
[root@pbx /]# systemctl enable openvpn@
Failed to execute operation: Unit name openvpn@.service is missing the instance name.
[root@pbx /]# systemctl enable openvpn
Failed to execute operation: No such file or directory
[root@pbx /]# systemctl enable openvpn-client@
Failed to execute operation: Unit name openvpn-client@.service is missing the instance name.
[root@pbx /]# systemctl enable openvpn-server@
Failed to execute operation: Unit name openvpn-server@.service is missing the instance name.


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@woodpecker505 wrote:

I was able to use XMPP on FreePBX 13 (latest version modules) for a moment, having entered 1 user. I used a BRIA 4 PC softclient and presence was working for some time. Then, after changing the user name and password I'm now always getting authentication errors. Same thing for other users that I create.

XMPP sent:
XMPP Protocol
AUTH [xmlns="urn:ietf:params:xml:ns:xmpp-sasl" mechanism="PLAIN"]
xmlns: urn:ietf:params:xml:ns:xmpp-sasl
mechanism: PLAIN
CDATA: AGhlcm1hbgBoZXJtYW4=

XMPP received:
XMPP Protocol
FAILURE [xmlns="urn:ietf:params:xml:ns:xmpp-sasl" condition="not-authorized"]
TEXT: Authentication failure
xmlns: urn:ietf:params:xml:ns:xmpp-sasl
condition: not-authorized

I tried reinstalling XMPP, have reinstalled pm2, but no success.

Is there any logging and configuration setting available ? I found out that prosody is no longer used - so I looked a while in the wrong direction, but now I have no clou what is going wrong. As said, it has worked for my first user in the beginning until I changed the name and password. My users have XMPP set in user manager, and password has been reentered afer reinstalling XMPP.

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@billsimon wrote:

I've been experimenting with the current distro and find that it won't install on a 512MB machine (the installer runs out of memory). I can install it with 2GB and then reduce the size of the VM to 512MB and it seems the services barely fit in this footprint, swapping a bit. My conclusion is that 1GB would be about the minimum memory allocation for any practical, small use. I don't see any minimum requirements posted. Could anyone comment further?

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@Loki wrote:

Hi all,

I am experiencing opportunistic crashes of asterisk approximately once a week without any significant log entry or resource peak preceding it. This started occurring when I upgraded to the 10.13.66-21 version. My versions of Asterisk packages:

asterisk13*.x86_64 13.17.0-2.shmz65.1.176

So I have a fix for the bug described here After upgrade to 10.13.66-21 asterisk is crashing

Core dump from last crash is here: Core dump

Any suggestion how to prevent these crashes is welcome.

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@WB3FFV wrote:

My local PBX here was acting goofy, and when looking under SIP peers I am seeing a lot of what looks like replicated extensions that start with 99 or 9999. I use three digit extensions, and say I have a 201 extension, I also seem to have a 99201 and a 9999201 in my sip peers.

Looking for it in the asterisk directory, I see they show in extensions_additional.conf and in sip_additional.conf, so it seems like something FreePBX would have generated, but I am not sure why.

Any ideas what is up here, or why I am seeing this? Or is my system compromised and I am just now finding it.

What started me looking is that about a day or so ago, calls are taking forever to start, incoming, or outgoing, so much so inbound calls often return no trunks available. I have yet to figure that out..

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@james wrote:

Hello：
I install FreePBX Distro-14, but the UCP link seems broken. I tried both link from login and ucp link from admin side.
this show error:

---

error:There was an error trying to load UCP:asort() expects parameter 1 to be array, null given.

any suggestion?

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@Loki wrote:

Hi,

with current distro (10.13.66) I have been experiencing asterisk crashes and deadlocks. Each time I created a backtrace, the developers of Asterisk told me it is optimized and missing debug symbols, and therefore it is of no use.

So my question is, has anyone of you created a backtrace on the FreePBX distro that was accepted by the asterisk developers? If so, how did you proceed?

I know asterisk is compiled with DONT_OPTIMIZE on the distro but yet, it was not enough. Do I need to recompile asterisk from source? If so, is it safe on the distro to run custom build asterisk? Do you have any experience with that?

Thank you for any opinion

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@paok1926 wrote:

hello all,

just installed STABLE SNG7-FPBX-64bit-1707-1 and i'm trying to connect this to my vpn server.
i installed pptp.x86_64 package, created the connection, all seems fine, but my connection fails every 1'.
i have a script to check for it and restart the connection.. below you can find the log file...

Aug 19 23:14:21 pbx pptp[9289]: anon log[main:pptp.c:333]: The synchronous pptp option is NOT activated
Aug 19 23:14:21 pbx pptp[9294]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 1 'Start-Control-Connection-Request'
Aug 19 23:14:21 pbx pptp[9294]: anon log[ctrlp_disp:pptp_ctrl.c:754]: Received Start Control Connection Reply
Aug 19 23:14:21 pbx pptp[9294]: anon log[ctrlp_disp:pptp_ctrl.c:788]: Client connection established.
Aug 19 23:14:22 pbx pptp[9294]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 7 'Outgoing-Call-Request'
Aug 19 23:14:22 pbx pptp[9294]: anon log[ctrlp_disp:pptp_ctrl.c:873]: Received Outgoing Call Reply.
Aug 19 23:14:22 pbx pptp[9294]: anon log[ctrlp_disp:pptp_ctrl.c:912]: Outgoing call established (call ID 0, peer's call ID 2129).
Aug 19 23:14:22 pbx pppd[9295]: pppd 2.4.5 started by root, uid 0
Aug 19 23:14:22 pbx pppd[9295]: Using interface ppp0
Aug 19 23:14:22 pbx pppd[9295]: Connect: ppp0 <--> /dev/pts/1
Aug 19 23:14:24 pbx pppd[9295]: CHAP authentication succeeded
Aug 19 23:14:24 pbx pppd[9295]: MPPE 128-bit stateless compression enabled
Aug 19 23:14:25 pbx pppd[9295]: local IP address 10.122.91.14
Aug 19 23:14:25 pbx pppd[9295]: remote IP address 10.122.91.13
Aug 19 23:15:00 pbx php: /sbin/ip6tables -A fpbxinterfaces -i ppp0 -j zone-trusted
Aug 19 23:15:00 pbx php: /sbin/iptables -A fpbxinterfaces -i ppp0 -j zone-trusted
Aug 19 23:15:02 pbx pppd[9295]: Terminating on signal 15
Aug 19 23:15:02 pbx pppd[9295]: Connect time 0.7 minutes.
Aug 19 23:15:02 pbx pppd[9295]: Sent 84 bytes, received 420 bytes.
Aug 19 23:15:02 pbx pptp[9791]: anon log[main:pptp.c:333]: The synchronous pptp option is NOT activated
Aug 19 23:15:02 pbx pppd[9295]: MPPE disabled
Aug 19 23:15:02 pbx pptp[9294]: anon log[pptp_read_some:pptp_ctrl.c:559]: read returned zero, peer has closed
Aug 19 23:15:02 pbx pptp[9294]: anon log[callmgr_main:pptp_callmgr.c:266]: Closing connection (shutdown)
Aug 19 23:15:02 pbx pptp[9294]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 12 'Call-Clear-Request'
Aug 19 23:15:02 pbx pptp[9294]: anon log[pptp_read_some:pptp_ctrl.c:559]: read returned zero, peer has closed
Aug 19 23:15:02 pbx pptp[9294]: anon log[call_callback:pptp_callmgr.c:81]: Closing connection (call state)
Aug 19 23:15:02 pbx pppd[9295]: Connection terminated.
Aug 19 23:15:02 pbx avahi-daemon[635]: Withdrawing workstation service for ppp0.
Aug 19 23:15:02 pbx pptp[9829]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 1 'Start-Control-Connection-Request'
Aug 19 23:15:02 pbx pptp[9829]: anon log[ctrlp_disp:pptp_ctrl.c:754]: Received Start Control Connection Reply
Aug 19 23:15:02 pbx pptp[9829]: anon log[ctrlp_disp:pptp_ctrl.c:788]: Client connection established.
Aug 19 23:15:02 pbx pppd[9295]: Exit.
Aug 19 23:15:02 pbx pptp[9300]: anon warn[decaps_hdlc:pptp_gre.c:217]: short read (-1): Input/output error
Aug 19 23:15:02 pbx pptp[9300]: anon warn[decaps_hdlc:pptp_gre.c:229]: pppd may have shutdown, see pppd log
Aug 19 23:15:03 pbx pptp[9829]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 7 'Outgoing-Call-Request'
Aug 19 23:15:03 pbx pptp[9829]: anon log[ctrlp_disp:pptp_ctrl.c:873]: Received Outgoing Call Reply.
Aug 19 23:15:03 pbx pptp[9829]: anon log[ctrlp_disp:pptp_ctrl.c:912]: Outgoing call established (call ID 0, peer's call ID 2130).
Aug 19 23:15:03 pbx pppd[9872]: pppd 2.4.5 started by root, uid 0
Aug 19 23:15:03 pbx pppd[9872]: Using interface ppp0
Aug 19 23:15:03 pbx pppd[9872]: Connect: ppp0 <--> /dev/pts/2
Aug 19 23:15:05 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 1 (expecting 24)
Aug 19 23:15:05 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 2 (expecting 24)
Aug 19 23:15:05 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 3 (expecting 24)
Aug 19 23:15:06 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 4 (expecting 24)
Aug 19 23:15:06 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 5 (expecting 24)
Aug 19 23:15:09 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 6 (expecting 24)
Aug 19 23:15:09 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 7 (expecting 24)
Aug 19 23:15:12 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 8 (expecting 24)
Aug 19 23:15:12 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 9 (expecting 24)
Aug 19 23:15:15 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 10 (expecting 24)
Aug 19 23:15:17 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 11 (expecting 24)
Aug 19 23:15:18 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 12 (expecting 24)
Aug 19 23:15:21 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 13 (expecting 24)
Aug 19 23:15:24 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 14 (expecting 24)
Aug 19 23:15:24 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 15 (expecting 24)
Aug 19 23:15:27 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 16 (expecting 24)
Aug 19 23:15:30 pbx pptp[9875]: anon log[decaps_gre:pptp_gre.c:418]: discarding duplicate or old packet 17 (expecting 24)
Aug 19 23:15:33 pbx pppd[9872]: LCP: timeout sending Config-Requests
Aug 19 23:15:33 pbx pppd[9872]: Connection terminated.
Aug 19 23:15:33 pbx avahi-daemon[635]: Withdrawing workstation service for ppp0.
Aug 19 23:15:34 pbx pppd[9872]: Modem hangup
Aug 19 23:15:34 pbx pppd[9872]: Exit.
Aug 19 23:15:34 pbx pptp[9875]: anon warn[decaps_hdlc:pptp_gre.c:217]: short read (-1): Input/output error
Aug 19 23:15:34 pbx pptp[9875]: anon warn[decaps_hdlc:pptp_gre.c:229]: pppd may have shutdown, see pppd log
Aug 19 23:15:34 pbx pptp[9829]: anon log[callmgr_main:pptp_callmgr.c:242]: Closing connection (unhandled)
Aug 19 23:15:34 pbx pptp[9829]: anon log[ctrlp_rep:pptp_ctrl.c:254]: Sent control packet type is 12 'Call-Clear-Request'
Aug 19 23:15:34 pbx pptp[9829]: anon log[call_callback:pptp_callmgr.c:81]: Closing connection (call state)

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@adtopkek wrote:

This isn't a question I am just documenting what I did to fix a problem while upgrading to freePBX 14 to help others and myself in the future. I didn't find this info while searching so maybe now someone else can find it.

I was running the freepbx 13 distro ISO. I ran the Freepbx 13 -> 14 upgrade script and ran into an apache problem along the way. It would not start.

service httpd restart
Redirecting to /bin/systemctl restart httpd.service
Job for httpd.service failed because the control process exited with error code. See "systemctl status httpd.service" and "journalctl -xe" for details.

I checked the journalctl

-- Unit httpd.service has begun starting up.
httpd[25309]: (98)Address already in use: AH00072: make_sock: could not bind to address [::]:80
httpd[25309]: (98)Address already in use: AH00072: make_sock: could not bind to address 0.0.0.0:81
httpd[25309]: no listening sockets available, shutting down
httpd[25309]: AH00015: Unable to open logs
systemd[1]: httpd.service: main process exited, code=exited, status=1/FAILURE
kill[25311]: kill: cannot find process ""
systemd[1]: httpd.service: control process exited, code=exited status=1
systemd[1]: Failed to start The Apache HTTP Server.
-- Subject: Unit httpd.service has failed
-- Defined-By: systemd
-- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
--
-- Unit httpd.service has failed.
--
-- The result is failed.
systemd[1]: Unit httpd.service entered failed state.
systemd[1]: httpd.service failed.

I found out that in /etc/httpd/conf/httpd.conf it was set to listen on port 80 as is expected:

# Change this to Listen on specific IP addresses as shown below to
# prevent Apache from glomming onto all bound IP addresses.
#
Listen 80

So the system was listening on port 80 as it should. The problem is that it was told to listen a 2nd time at /etc/httpd/conf.d/schmoozecom.conf

# Automatically Generated File
# Starting acp
# acp
Listen 80
< VirtualHost *\:80>
....

All I did was comment out the listen in /etc/httpd/conf.d/schmoozecom.conf and it worked after a "service httpd restart"

# Automatically Generated File
# Starting acp
# acp
#Listen 80
< VirtualHost *\:80>
.....

Hope this helps someone else!

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@thehammer86 wrote:

Hi,

I have my FreePBX 14 Asterisk 13 box set up for IPv6. SSH over IPv6 works as well as web admin and UCP.

I get errors when trying to connect via IPv6 using extensions set up with CHAN-SIP as it says the address family is not supported. No problem, so I went to set up PJ-SIP using IPv6.

I added my entire /56 subnet to the firewall trusted list to start. Responsive firewall is enabled for PJ-SIP and in the services section PJ-SIP is set to allow local as well.

My extensions, when forced to IPv6 mode only, can't seem to connect at all. There are no entries in the PBX logs.

I have TLS set up for CHAN-SIP and PJ-SIP and SRTP which all works over IPv4.

I noticed in pjsip.transports that there is no bind=IPv6 address and only a bind=IPv4 so I added the NIC's address in as bind=[XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX]:5062. That didn't seem to work so I also tried bind=XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:5062 since my polycom vvx300 reported a connection error in that format, but still nothing.

Anyone at Sangoma or the community have a working IPv6 step-by-step guide for PJ-SIP?

Thanks in advance!

P.S. Here is my ifcfg-eth0 file:

DEVICE=eth0
BOOTPROTO=static
ONBOOT='yes'
IPADDR=XXX.XXX.XXX.XXX
NETMASK=255.255.255.248
GATEWAY=XXX.XXX.XXX.XXX
ZONE=external
DESCRIPTION="unset"
IPV6INIT=yes
IPV6ADDR=XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX:XXXX/128
IPV6_MTU=1492
DEFROUTE=yes
IPV6_AUTOCONF=yes
IPV6FORWARDING=no
NM_CONTROLLED=no

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@adtopkek wrote:

I upgraded a backup of one of our servers from 13 -> 14. We were using the 13 distro originally and the upgrade went smooth until it came to module updates and the http server, http got fixed.

Upon doing module updates the only modules that would properly update was Framework and I had to specifically upgrade sysadmin as it said broken but that got updated as well. It was trying to download core and was saying this:

fwconsole ma update core
No repos specified, using: [standard,extended,unsupported,commercial] from last GUI settings
Downloading module 'core'
The following error(s) occured:
- Retrieved Module XML Was Empty

When running an "fwconsole ma listonline" it showed this:

Selection_374.png689x817 232 KB

Now I was able to progress after banging my head against a wall for long enough. I found these topics which led me to a partial fix:
https://issues.freepbx.org/browse/FREEPBX-9021

After emptying/deleting /etc/schmooze/pbx-brand I was able to update about 75% of my modules.

The remaining modules say "broken" and when trying to update I get:

fwconsole ma update parkingpro
No repos specified, using: [standard,extended,unsupported,commercial] from last GUI settings
Downloading module 'parkingpro'
The following error(s) occured:
- Retrieved Module XML Was Empty

I was able to install an update for endpoint after a reboot but the rest of them I haven't been able to update.

Broken

Module Version Publisher License Status Track
extensionroutes 13.0.10.4 Stable Broken
pagingpro 13.0.19.6 Stable Broken
parkpro 13.0.30.10 Stable Broken
vmnotify 13.0.18.3 Stable Broken
voicemail_report 13.0.13.2 Stable Broken
recording_report 13.0.24.6 Stable Broken
restapps 13.0.88.5 Stable Broken
conferencespro 13.0.27.7 Stable Broken

Any idea what I can do to get these remaining modules updated? Freepbx and Asterisk work for the most part right now besides those broken modules.

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@eggythetech wrote:

I have a 14 Sangoma PBX 60 with FreePBX 1707-1 Distro on them.
When i reboot the Nic does not Start. i have verified that it is set to start on boot.
If i Statically set the IP and reboot it still does not start.
If i do a fwconsole restart then the nic will initialize.
any idea if its a bad Build on the distro? or any suggestions would be great.

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@wa4zlw wrote:

when will the latest security patches be pushed out?

thanks leon

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@rn0801 wrote:

Installed FreePBX, applied all updates. Trunks and SIP-phones are all PJSIP.

SIP-part is fully functional.

SANGOMA A500 is installed as bri_net_ptmp to connect internal ISDN-Phones.

All defined DAHDI-extensions can be dialled (from SIP-extensions and from the outside).

When I dial from a DAHDI-extension, the dialplan only receives the CID of the calling extension but no dialled number at all.

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@thehammer86 wrote:

Hi,

Since I've upgraded to FreePBX 14 Stable Asterisk 13 almost two months ago, I've noticed perioidc crashes of asterisk. I ignored them because they were't that frequent and FreePBX 14 just became stable so I was waiting to see if it may be a bug that would be eventually ironed out.

At the time of Asterisk crashing this appears in the freepbx.log file:

[2017-Sep-25 16:28:00] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:00] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:01] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:01] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:01] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:01] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:01] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:01] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:01] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:01] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 58
[2017-Sep-25 16:28:02] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
[2017-Sep-25 16:28:02] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:02] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:02] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:03] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:03] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:03] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:03] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:03] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:03] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:04] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed
[2017-Sep-25 16:28:04] [INFO] (restapps/restapps.php:55) - Phone Applications daemon started.
[2017-Sep-25 16:28:04] [CRITICAL] (admin/bootstrap.php:270) - Connection attmempt to AMI failed

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@matthewljensen wrote:

I recently set up a couple of remote extensions and I've been experiencing problems with them. My server is hosted elsewhere, so technically all of my extensions are remote, but these new ones have been especially troublesome, while at our main office the phones work perfectly.

I'm not currently posting any technical information because I'm not sure what is needed to diagnose this issue. I can update with logs or other technical information with guidance as to what I should post.

Here is what is happening:

I can make calls with these extensions, either to other extensions or to an unrelated phone, and the other party receives the call. They can answer, and they show as connected. But only sometimes does sound go through, mostly it does not. It seems rather hit and miss. There is a possibility that making a call after waiting a long time seems to have more success, but I'm not sure if this is the case. Also, sometimes after waiting for around 20-30 seconds, the call will "connect" again, and sound will go through. But again, this is not always the case. BLF info is coming through as well. And they are listed as OK in the astierisk infos peers section.

I have tried changing sip settings to nat = yes.
I've tried temporarily disabling iptables, even though our IPs are whitelisted here. (I'm using travelin man)

One thing that may be the problem is that we bounce between a couple of public IPs rather frequently. Could that be the problem?

I'm tripped up because this problem seems so hit and miss. Do you guys have any suggestions? I can post more information here if it is needed, but what I really need is another place to look for the problem.

Thanks so much.

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@adtopkek wrote:

I've been working with the default Freepbx 13 (responsive) firewall and when you connect to an extension directly using the server IP then the connecting device's IP address can access the UCP. That is a great feature and I am trying to get that to work with OpenVPN.

I setup openvpn through Freepbx, I formatted the OpenVPN file how my Yealink wants it, my Yealink connected to the VPN server, and calls out through the VPN. The problem is that the public IP I am using to connect to the OpenVPN server cannot connect to the UCP as it is not being allowed through like if I was directly registering my endpoint without using the VPN tunnel.

Ip 81.35.27.26 connects to 10.8.0.1 and gets the IP 10.8.0.2. It uses 10.8.0.1 to talk to the Freepbx server to avoid firewall issues. I want 81.35.27.26 to be able to access the UCP.

How do I get IP's using the OpenVPN to be able to access the UCP automatically?

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@thehammer86 wrote:

Does anyone happen to know why the following message would be appearing in my logs randomly?

[2017-09-27 07:51:48] ERROR[28249] pjproject: ssl0x7fd6ac12fbb0 Renegotiation failed: Not enough memory (PJ_ENOMEM)

My server has 8GB of memory and more than 60% remains unallocated on average. Also, there was no obvious memory spike at the time of that log entry.

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@adtopkek wrote:

I need to change a few settings on the Distro OpenVPN server. As noted:

Configuration automatically generated via Sysadmin RPM
MODIFICATIONS TO THIS FILE WILL BE OVERWRITTEN.

when I changed something and restarted it the settings were over written. Where do I go to permantly change some of the .conf settings?

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