Hi Guys I am in the office today and here is what I got from AT&T
Hi Arturo, Sadly I don’t actually remember if we got that work or not. I know our customer service managers liked using the FreePBX gui instead so I’m not sure if it ever actually worked or not. Our call center has since moved off-site so I don’t even have a way to go back and test anymore. I might setup a test vm and check inside of there just for the sake of it though. I’ll let you know what I turn up.
I am new to the Freepbx environment but have found some really good help in this forum, I am currently working on upgrading our phone system from 2.11 on Cent OS 6.7 to Freepbx 14 Distro SNG7 Official, I utilized the Distro Conversion Tool (Elastix to PBXinaFlash to FreePBX Distro) have new server imaged with freepbx 14 and transferred everything over from the donor machine everything looks good. But as far as the cutover to the new system not sure what else is required, I know that I will have to swap IP Addresses for the Registar Server or should I just run them side by side and start transitioning people over to the new machine with new IP Address. Our environment isn’t that complicated we have about 50 Extensions with about 20 inbound routes and some daily schedules for call routing. Please help.
The conversion tool does not copy over any custom stuff.
I also found that some misc applications/destinations were disabled.
All Trunks are disabled.
See here how copy some stuff that is not being copied over with the tool: How to: Upgrade FreePBX13 to FreePBX14 on Hyper V - Guide
As far as the cutover, I’d just change IPs, so in case something is badly wrong on the new PBX you can just switch back
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we fixed this on 2 systems. here are the notes.
innodb-huge-solution.tgz (1.1 KB)
Anyone else having this issue? Or perhaps someone has a suggestion as to where to look next?
That code means the call went into the first IVR, you see each IVR gets a number, your first IVR appears to me misconfigured therefore the call never gets terminated and/or is going into a loop, check the IVR settings, the tim out settings and the return setting should be set in a way that they dont keep on looping also make sure to set return after VM to “NO” if you revise the IVR settings I am sure you will find the issues.
Good Luck
really? notes in a tar file?
I took a moment to open that in my disposable VM for the rest of us.
* phonesys mysql issue .
- make snapshot in proxmox [ backup up the system ].
getting alerts of low disk space. <<< change made see below <<
** keep track of changes, perhaps file maxed out?
$ sudo ssh sip "ls -l /var/lib/mysql/ibdata1"
-rw-rw----. 1 mysql mysql 6520045568 Jul 29 15:01 /var/lib/mysql/ibdata1
-rw-rw----. 1 mysql mysql 6528434176 Jul 29 17:14 /var/lib/mysql/ibdata1
-rw-rw----. 1 mysql mysql 6528434176 Jul 30 09:04 /var/lib/mysql/ibdata1
https://vdachev.net/2007/02/22/mysql-reducing-ibdata1/ at Dump the Whole Database
mysqldump --extended-insert --all-databases --add-drop-database --disable-keys --flush-privileges --quick --routines --triggers > '/tmp/all-databases.sql'
# trying to elim 'Firewall was unable to connect to MySQL after 30 seconds.'
systemctl stop crond
systemctl stop sangoma-pnpd.service
systemctl stop incrond.service
systemctl stop mariadb.service
# Note we kept phone system on duiring the long process. prob only affected logging.
#kill
# ps -ef | grep firew | grep -v grep
root 8039 1 0 Jul20 ? 00:08:30 php /var/www/html/admin/modules/firewall/hooks/voipfirewalld
root 8580 8039 0 Jul20 ? 00:50:16 voipfirewalld (Monitor thread)
#-----------------------------------------------------------------------------
nano /etc/my.cnf
# set these:
innodb_file_per_table=1
innodb_flush_method=O_DIRECT
innodb_log_file_size=250M
innodb_buffer_pool_size=1G
#-----------------------------------------------------------------------------
mv /var/lib/mysql /var/lib/mysql.ori
# May need to rm -fr /var/lib/mysql.ori if space low. check with df
mkdir /var/lib/mysql
chown mysql:mysql /var/lib/mysql
sudo -u mysql mysql_install_db
systemctl start mariadb.service
mysql
SET FOREIGN_KEY_CHECKS=0;
SOURCE /tmp/all-databases.sql;
SET FOREIGN_KEY_CHECKS=1;
systemctl restart mariadb.service
systemctl start crond
systemctl start sangoma-pnpd.service
systemctl start incrond.service
I rebooted to make sure firewall was back on. TBD how to start it.
rm -f /tmp/all-databases.sql
For the record, I decided it was an insane amount of storage for “dashboard stats” and simply dropped and recreated the kvblobstore table.
I used mysqldump to create a backup.sql and then took the table creation part of kvblobstore out of it and left it as a .sql file to run.
mysql -uroot asterisk < ~/kvblobstore.sql
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I buy the Sangom S305 phone + Sysadmin Pro module.
I setup the OpenVPN and confirm from my laptop it connects fine and I can access the FreePBX
Then I provision the phone for OpenVPN and the VPN icon shows up at time. But I take the phone offsite and the VPN connection is not established, it shows in the menu VPN IP = 0.0.0.0
How can I view the logs of VPN on the phone? When I login to the web interface there is no mention of VPN
I currently have a working system - FreePBX 12 and many Polycom IP phones.
Right now, everything is on the default VLAN 1 (192.168.1.x).
For testing purposes, I moved 1 phone to a different VLAN 12 (192.168.12.x). At this point, communication between VLAN 1 and 12 is wide open. There are no restrictions and everybody can communicate with everybody else.
When I boot the test phone on VLAN 12, it grabs the right IP address, it can connect to the provisioning server, grabs the correct settings, it gets the phone background, and it grabs the current time. The only thing it will not do is register the line.
I can ping the phone from VLAN 1 and even connect to the phones web interface. I hooked up a laptop on VLAN 12 and I can ping the FreePBX server and login to the admin interface.
I have 192.168.1.x and 192.168.12.x listed as local networks under Settings -> Asterisk SIP Settings -> General SIP Settings -> NAT Settings -> Local Networks
If I switch the phone back to VLAN 1, the line registers fine.
What am I missing?
I don’t think that person is stuck with this issue for 5 years now 
Might have got banned under intrusion detection.
If it is, whitelist 192.168.12.0/24 and restart the service.
If not, what do you see in the logs when it attempts to register?
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We’ve been over why this doesn’t work.
Also the “did you get this to work?” and “do you want my configs?” on a three year old question is kind of pointless. Is it still relevant now? As in, is the OP still running FreePBX v12 ? Which version of Asterisk is it running? Since it’s v12 that means 10/11/12 of Asterisk.
Also in May 2016, FreePBX v13 (along with Asterisk 13) was the main focus with v12 being retired. So even then the answer might have been to upgrade to a more current version of FreePBX.
FreePBX v14 on SNG7 is nothing like FreePBX v12 on CentOS 5. There’s been a complete overhaul of FreePBX between 12 and 14 (even 12 and 13). Numerous updates have been done, bugs fixed, new features released that may have deprecated old features.
Also, if you read this tread almost no real debugging or troubleshooting was done. The last reply was the OP being told that they should look at Digum’s stuff for answers and to message someone who seemed to know Digium stuff well. Not sure any of that was done.
I’m not sure why 2019 has seen a trend of people reviving 2-3+ year old posts over an issue that is 90% likely to not even apply to current deployments because, updates, version changes and all that. A FreePBX system setup in 2015 will not be the same as a FreePBX system setup in 2019.
I am having an issue with my configuration. I am not able to call to external numbers.
I am using asterisk 13 with freepbx 14. I am using pjsip configuration.
I did some digging with “pjsip show history” and I found that my server sends the invitation to the sip server but after “trying” there is a message that says “502 bad gateway”. The message says:
SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;received=XXX.XXX.XXX.XXX;branch=
From: “XXXXXX” sip:XXXXXXXX@XXX.XXX.XXX.XXX;tag=
To: sip:XXX.XXX.XXX.XXX@XXX.XXX.XXX.XXX;tag=gyodtxgc
Call-ID:
CSeq: 5465 INVITE
Warning: 399 37959.2432.P.261.5.105.0.4.32839.0.1075445762.ims.MY.PROVIDER.COM “Service lost”
Content-Length: 0
Content-Length: 0
The thing is that I did not change anything in my network configuration. Now it is working in a virtual machine and I have a snapshot with everything working, but I can not reload the configuration to make any change.
Thanks for your help
This is a call your provider issue.