I am looking for best upgrade path. We have a dedicated physical machine for a small medical office. 15 or so internal extensions. 1 trunk. Small simple installation, works great.
Right now we are running on: 5.211.65-16 and I'd like to update us to the latest version and also migrate us to a virtual machine on our VMWare installation. From what I have read for a small installation like this it should be fine on virtual hardware. No telephony hardware.
I would like to do a clean installation on the VM of the FreePBX distribution. However, I've read that you cannot do a restore from a previous version. The backups created are version specific.
What is my best upgrade path or plan here? Should I take the current physical machine and run through the upgrade script bringing it to current, make a backup and then restore it to the new installed machine of the same version?
I also have commercial modules (endpoint, REST Apps). How will those get moved over to the new installation?
In CDR Reports there are some filters like Not |_| Begins With: Contains: Ends: With: Exactly:
Everything works fine with CallerID Number,CallerID Name,DID, e.t.c But in Destination field checkpoint 'Not' doesn't work.
As example: I have 2 inbound calls 1. CallerID 555-555,Destination 101 2. CallerID 555-555,Destination 102
I can use filter 'Destination Contains: 101' and I see just one call, I can use filter 'CallerID Not Contains: 555-555' and I see no calls. But then I use filter 'Destination Not Contains 101' i still see 2 calls.
Trying to install FreePBX on an Acer Aspire R11 Notebook and not having much success. Trying to install via USB and CD.
I can't get the .img files to work at all using Win32DiskImager. That just seems to create a USB stick I can't read in Windows and doesn't boot in UEFI or Legacy mode.
Had some success with unetbootin and iso2usb, by changing from UEFI to Legacy, however I'm getting a kickstart error that it can't find the asterisk****.cfg file.
Finally, I bought a USB CD Rom and burnt the iso to DVD. Again I have to change from UEFI to Legacy to get it to boot, and this time it gets to the point where i goes to find the hard drive to install, and it says it can't find any usable disks. I'm wondering if this could be because it's formatted with GPT instead of MBR? Any recommendations of what try next?
Is there a way to install the Schmooze Commercial Module Installer manually? I have a FreePBX13/Asterisk13 system in Ubuntu 14.04 that I built manually (not from a Distro) but want to add commercial modules such as XMPP Pro.
I really...REALLY.. don't want to have to re-install the whole platform from a Distro. I'd much rather retain control over what is installed and understand how everything has been fitted together..especially as everything is working seamlessly.
I have a new Asterisk / FreePBX server using * CentOS 7 * Asterisk 13.6.0 * FreePBX 12.0.76.2
I am unable to upgrade modules from Module Admin or from the command line. After much investigation, I see that I get "500 Internal Server Error" as a reply to the download POST request in modulefunctions.class.php:1366. I see that there is a long and complicated POST string (quoted below). If I remove that string from the request the download succeeds.
Please can you help me understand why the POST string causes your server to fail?
Error! close Reload failed because retrieve_conf encountered an error: 255
exit: 255 Added to globals: ASTETCDIR = /etc/asterisk Added to globals: ASTMODDIR = /usr/lib/asterisk/modules Added to globals: ASTVARLIBDIR = /var/lib/asterisk Added to globals: ASTAGIDIR = /var/lib/asterisk/agi-bin Added to globals: ASTSPOOLDIR = /var/spool/asterisk Added to globals: ASTRUNDIR = /var/run/asterisk Added to globals: ASTLOGDIR = /var/log/asterisk Added to globals: CWINUSEBUSY = true Added to globals: AMPMGRUSER = admin Added to globals: AMPMGRPASS = amp111 Added to globals: AMPDBENGINE = mysql Added to globals: AMPDBHOST = localhost Added to globals: AMPDBNAME = asterisk Added to globals: AMPDBUSER = asteriskuser Added to globals: AMPDBPASS = astpassWD Added to globals: VMX_CONTEXT = from-internal Added to globals: VMX_PRI = 1 Added to globals: VMX_TIMEDEST_CONTEXT = Added to globals: VMX_TIMEDEST_EXT = dovm Added to globals: VMX_TIMEDEST_PRI = 1 Added to globals: VMX_LOOPDEST_CONTEXT = Added to globals: VMX_LOOPDEST_EXT = dovm Added to globals: VMX_LOOPDEST_PRI = 1 Added to globals: MIXMON_DIR = Added to globals: MIXMON_POST = Added to globals: DIAL_OPTIONS = Ttr Added to globals: TRUNK_OPTIONS = Tt Added to globals: TRUNK_RING_TIMER = 20 Added to globals: MIXMON_FORMAT = wav Added to globals: REC_POLICY = caller Added to globals: RINGTIMER_DEFAULT = 15 Added to globals: TRANSFER_CONTEXT = from-internal-xfer PHP Fatal error: Uncaught exception 'Exception' with message 'Critical error when splicing into macro-dialout-trunk. I was asked to splice into an empty section with a priority greater than 1. This is always a bug in a module. I was asked to add null' in /var/www/html/admin/libraries/extensions.class.php:233 Stack trace:
thrown in /var/www/html/admin/libraries/extensions.class.php on line 233
I have deleted all trunks one by from freepbx than i clicked "Apply Config" but after that i got and error like above. What is the reason of this error? And how can i fix this error?My freepbx version is 12.0.7
I have been working in Basefile Edit for the last few days in Endpoint Manager. Today there was a Module Update to 13.0.15, and now I there are no menu options to add lines while in Basefile Edit. This latest update to Endpoint Manager took away the option to add lines.
Hi all, if possible, please add this comment to the following post: modul-update-problems/19455
I solved this problem that came up around in December 2015 for me by changing the true from wget to false which was mentioned in this post the other way round.
Hope this helps someone who has a similar problem.
I am having issues in deleting Job Names that I have created in Backup & Restore. They will not delete, just like to know how I can delete the Job Names? I am using version 13.0.21.2. Any help would be appreciated.
I have a fresh Asterisk/FreePBX install configured to use a Grandstream GXW4104 gateway. The Gateway is configured for 2-stage dialing. When I use 1-stage, I get an "All circuits are busy". Inbound calls work and are routing properly. Outbound calls are half-way working. When I dial an outbound number 9XXXXXXX, the call seems to be transferred to the gateway (slight ring), then the outside line is picked up and I hear the dial tone, but it does not dial. However, once the outside line is picked up, I can then redial the 7 digit number and the call continues. I hear both sides of the conversation and the call quality is good.
I've been looking at these configurations for the past few hours and nothing. I'm not sure what side I need to be looking on. Here are some specific questions, but feel free to add any additional comments.
1) I think the issue is the PBX sending the DTMF tones, but not sure. I've starting going down this rabbit trail to see where it will lead! Am I looking in the right place?
2) On the gateway, should I be using 1 stage or 2-stage dialing? I think I should be using 1 stage, but keep getting the "all circuits are busy". But two stage dialing works, but I have to dial the number again once the line is picked up.
htop is showing /var/www/html/admin/modules/dashboard/scheduler.php using 100% cpu on one core of the eight that are available (2 CPUs, 4 cores each). This is even after an fwconsole restart and a full reboot. Not that restarting should really fix anything.
I recently installed FOP2 which seems to go along with the timing of when this started but not totally sure.
This looks like something running from the FreePBX dashboard but it seems to be working fine on the web page loading and status. Nothing else out of the ordinary that I can tell.
Please help me to solve the problem below - My PBX System : FreePBX 2.8 with Asterisk 1.8. - My Customer Care Group need setup a Queue with Only a Dynamic Agent : + in addition to the usual features, when : + The Agent = Busy (all agents in call) : asterisk play a announcement " All agents is Busy now , please wait a moment".. After that asterisk will tranfer the caller to a Ring Group. + The Agent = Unanswer (no Agent answer, not wait to timeout) : asterisk play another announcement " All agents i Available now, please try call again later " . After that asterisk will hangup call.
Can i configure my FreePBX (ver 2.8) for this inquiry or have to update FreePBX to solve ? Or anything else?
I'm trying to edit /etc/amportal.conf to connect freepbx with remote mysql database. The problem is when I edit the file and do 'amportal restart' the file return to options that were before edition.
Hi Guys, I've just completed a new build of FreePBX, Asterisk and CentOS 6.7, the server is remote to the building.
My status shows:
`SYSTEM INFORMATION
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE
Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE
SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A
---------------------------------------
PIAF Installed Version = 3.0.6.7 under *KVM*
FreePBX Version = 12.0.76.2
Running Asterisk Version = 12.8.1
Asterisk Source Version = 12.8.1
Dahdi Source Version = 2.10.0.1
Libpri Source Version = 1.4.15
IP Address = **.***.**.*** on eth0
Operating System = CentOS release 6.7 (Final) ><
Kernel Version = 2.6.32-573.8.1.el6.x86_64 - 64 Bit`
My handset is Cisco 7940G running sip firmware. The handset is accessing the server over the internet as it is downloading the configuration files etc from the tftp server however it is refusing to connect.
Within Asterisk I've got an extension set up called 101.
My phone cnf file is # Line 1 Settings line1_name: "101" ; line1_displayname: "101" ; line1_shortname: "101" ; line1_authname: "101" ; line1_password: "*********" ; proxy1_address: "***.**.***.**" ; proxy1_port: "5061";
How can I work out why my handset is failing to connect to my remote pbx?
Hello all. I have a problem with audio issues con my the company PBX system. I have a FreePBX 2.11.0.11 with Asterisk Ver. 11.4.0. I have a dedicated Dell Server for this. My setup is like this. I have a Quintun Tenor DX with 4 PRI T1's that is connected directly to my FreePBX server. I have my own VLAN for the phone's system and QoS on all the Cisco Switches on the company. We are not using SIP trunking. We have the Quintum configured like a SIP Trunk like this: PEER Details: username=xxxxxxxx type=peer secret=xxxxxxx insecure=very host=x.x.x.x disallow=all canreinvite=no canredirect=no allow=ulaw qualify=yes keepalive=45
We can make and receive call all the time. We have the issue that sometime we get words that repeats itself (Example: is everything Okay, Okay, Okay) Sometimes the call comes back and sometimes it stays on the loop. The other thing is we get audio drops on call for a couple of seconds, when this happens we are the party that does no here the other end. The other end always get the audio I have recorded some calls on the PBX itself. And the call are perfectly fine. So I assume that from the Quintum to my PBX is OK. and the problem is the audio from my PBX to my Phones. I have various types of phone:Cisco SPA504G, CP-7971G, Polycom IP 331 IP, 601 , IP 335 with the later SIP firmware. I have tested the latency from the FreePBX to some of the phones and I'm getting: icmp_seq=3 ttl=64 time=0.383 ms icmp_seq=11 ttl=64 time=0.373 ms and another phones I get: icmp_seq=3 ttl=64 time=0.405 ms icmp_seq=11 ttl=64 time=0.515 ms Intervals So is less that 1ms.
Is there a way to edit the voicemail folders? I would like to either edit the names of existing ones, or perhaps just create new ones and remove some of the existing ones.
I am running into an issue where if I give a user access to additional extensions in UCP, they lose access to their primary extension. Is this expected behavior? I note that with groups, there is the option to add "User Primary Extension" to the list (included by default, I believe), but this option is not available when editing a user. It isn't terrible to work around it and explicitly add the user's extension to the list, but that also means that if the user's primary extension changes, the permissions don't automatically follow.
Am I missing something here? Is this a bug, or is it the expected behavior?
exit: 255 Unable to continue. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'asterisk.freepbx_users' doesn't exist in /var/www/html/admin/modules/contactmanager/Contactmanager.class.php on line 604
