Running FreePBX distro, 6.12.65 as a virtual machine.

I'm having a heck of a time with the audio on this machine. Around the beginning of the year or so (can't pin it down due to lack of activity around the holidays), we started getting some serious 'chop' in our audio, especially in conferences (though it is present in every call, it's almost impossible to use a conference room due to the amount of audio chop).

I investigated our UDP traffic, and had our network guys do extensive testing on the network and internet connection to make sure we weren't just dropping packets due to a bad piece of network hardware. Finally did a dahdi_test, and found this:

Opened pseudo dahdi interface, measuring accuracy...
99.997% 99.913% 99.809% 99.997% 99.945% 99.750% 99.809% 99.907%
99.901% 99.939% 99.927% 99.907% 99.900% 99.858% 97.374% 99.947%
99.995% 99.903% 99.656% 99.546% 99.995% 99.998% ^C
--- Results after 22 passes ---
Best: 99.998% -- Worst: 97.374% -- Average: 99.771525%
Cummulative Accuracy (not per pass): 99.867

Pretty bad. I installed a brand new VM with similar settings and a fresh copy of freepbx, latest version. Same thing. I put it on a different host within our organization, same thing. If I go to another, similar host at a different company, they have a perfect dahdi_test.

So it's probably something in our VM configuration, I concluded. Me and our ESX guy tried increasing the priority given to the VM by setting the 'Latency Sensitivity" to "High" and reserving 2 full cores and 3gb memory, dedicated them to the PBX. It made absolutely no difference; in fact, I got worse results occasionally.

So we decided to get a Sangoma USB VoiceTime device; plugged it in and attached to the VM. I had to download and compile an older version of Dahdi (2.5 or so) in order to get the voicetime drivers to compile and install/work. I had to contact support to make sure the device was installed and working. They verified that it was.

Installing the VoiceTime device helped a little bit, but the same symptoms are still there and I have no idea what to try next. Help!

Oh, also, I lied; the dahdi_test results above were done about 10 minutes ago with the voicetime device attached. Unattached, it's pretty similar, but actually worse.

It's just a quick discussion and verification.

in my case, I have a more then capable cisco router and cisco switches handling the package tagging and routing.
Does having the FreePBX server have any decremented impact on overall performance being on separate network then the VoIP phone.
I also currently have the EPM module, which does a good job of provisioning and DHCP services showing where the server is to the phones.

The system is setup currently in production, just wanted to confirm keep this or switching to a flat network with server on same segment as the phones would be beneficial.

We just switched our VOIP line to SIPStation and ever since we made the switch we are no longer getting the CID (CNAME) information. With my old provider I could see the information with no issue but since the switch to SIPStation we can not. We have configured the SIPStation trunk using the SIPSTATION module and have a very simple configuration.

We are currently running FreePBX 13.0.51 / Asterisk Version: 13.5.0.

Hi,

I am attempting to install FreePBX on a VPS and I've followed the official guide for an install on a fresh CentOS 6.5 machine found here:

I am installing on a x86 machine and had to adapt the script accordingly, most notably by removing the references to " --libdir=/usr/lib64" but also to solve the problem of missing libjansson.so.4 - i did this by running the following lines before I installed freepbx:

echo "/usr/local/lib" > /etc/ld.so.conf.d/usr_local.conf
/sbin/ldconfig

I still had to run the installation twice as it hangs at first, but then goes through successful and I can access the control panel.

However, I immediately get an error that there is 1 Critical error with a tampered file:

"Critical Module "FreePBX Framework" is unsigned, re-download immediately"

But I am unable to re-download this module or run any update, I always receive an error message stating:

"Downloading and Installing framework
Downloading framework 0 of 5081756 (0%)
Found module locally, verifying...Redownloading
Error(s) downloading framework:
Error opening http://mirror1.freepbx.org/modules/packages/framework/framework-12.0.76.2.tgz.gpg for reading"

I also try to run "amportal chown" and "amportal a ma refreshsignatures" and again receive the following error:

Checking Signatures of Modules...
Checking builtin...Good
Checking callrecording...Good
Checking cdr...Good
Checking core...Good
Checking customappsreg...Good
Checking dashboard...Good
Checking featurecodeadmin...Good
Checking framework...Signature Invalid
Refreshing framework
Downloading 0 of 5081756 (0%) The following error(s) occured:
- Error opening http://mirror1.freepbx.org/modules/packages/framework/framework-12.0.76.2.tgz.gpg for reading

I can access the file with wget without any problem.

I've been around this for a few days already, would really appreciate any help.

Thanks in advance,

S

Hello Everyone,

I wanted to update you with what I ended up doing to solve this problem. Using FreePBX 13, I created a virtual extension, and used Follow Me within it to ring with an Alert-Info of "visual". Interestingly, the "visual" was not a selected option.... I had to type it into the field in order to make it stick.

I also found it handy to change the Follow Me destination to Voicemail of the "real" destination, so that the design would be complete, and that an an un-answered call would be properly dealt with.

I also discovered what the Account1 and Account2 were on the phones. I do wish the EndPoint Manager was better documented on how to use it properly.

Thank you everyone for your suggestions.

Christian

If you can open up a ticket to FreePBX support as well as give them a few examples of numbers that are calling you they should be able to investigate what is going on.

@andmore
just a quick follow-up. I don't know if you reported this or someone else, but support has informed me that the issue has been resolved. There was a configuration change in the SSL security that is used by SIPStation to do lookups of the CNAM from the provider which was resulting in broken CNAM unless SIPStation could fall back to other sources which don't have full coverage.

That has been addressed and you should be getting CNAM.

Thanks Rob,

I upgraded to FreePBX 13 and once I confirmed everything was ok I then followed the steps in your walkthrough and was able to successfully upgrade to Distro 6.6, all seems to be working fine...

Thanks!

Hi Andrew!

Sorrrrrry for the delayed reply...

I still don't quite understand why it took a few restarts for everything to get back to normal without me seemingly doing anything different from one time to the next but once it started working it never stopped working...

My DAHDi helper/config module is currently active and the configuration is done from FreePBX.

My guess is that it had to do with the permission aspects of the problem but I don't recall doing anything which could have changed them.

Thank you very much for your help in resolving this and have a nice day!

Nick

Current system configuration:
CentOS Linux 7 x86_64 AMW EC2 AMI
Apache v2.4.6
FreePBX v13 (Successfully Installed based on instructions: http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7#InstallingFreePBX13onCentOS7-Disableselinux)

Installation went without any hiccups however I am now having issues accessing the FreePBX v13 GUI. I am getting a “Forbidden You don't have permission to access /admin” on this server error. When running fwconsole debug this error stands out:

[Wed Jan 27 02:24:21.462454 2016] [core:crit] pid 2710Permission denied: [client xx.xxx.xx.xxx:xxxxx] AH00529: /var/www/html/admin/.htaccess pcfg_openfil e: unable to check htaccess file, ensure it is readable and that '/var/www/html/ admin/' is executable
[Wed Jan 27 02:24:31.214694 2016] [core:crit] pid 2711Permission denied: [client xx.xxx.xx.xxx:xxxxx] AH00529: /var/www/html/admin/.htaccess pcfg_openfil

See link for current configuration and more details: https://codeshare.io/h7Cpv

Andrew Nagy, I missed that.. so I reran it.. maybe I need to delete some things and then run it..

Should I be dumping the databases I made and deleting some files in etc ? You will note at bottom I still get error. but like I am guessing its probably because its not a clean install

` root@asteriskdebian:/usr/src/freepbx# ./install
Database engine [mysql]:
Database name [asterisk]:
CDR Database name [asteriskcdrdb]:
Database username [root]:
Database password: ***********************
File owner user [asterisk]:
File owner group [asterisk]:
Filesystem location from which FreePBX files will be served [/var/www/html]:
Filesystem location from which Asterisk configuration files will be served [/etc/asterisk]:
Filesystem location for Asterisk modules [/usr/lib/asterisk/modules]:
Filesystem location for Asterisk lib files [/var/lib/asterisk]:
Filesystem location for Asterisk agi files [/var/lib/asterisk/agi-bin]:
Location of the Asterisk spool directory [/var/spool/asterisk]:
Location of the Asterisk run directory [/var/run/asterisk]:
Location of the Asterisk log files [/var/log/asterisk]:
Location of the FreePBX command line scripts [/var/lib/asterisk/bin]:
Location of the FreePBX (root) command line scripts [/usr/sbin]:
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
Reading /etc/asterisk/asterisk.conf...Done
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Done
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...No (/etc/amportal.conf file detected)
Initializing FreePBX Settings
Finished initalizing settings
Copying files (this may take a bit)....
0/6165 [>---------------------------] 0%/etc/asterisk/manager.conf has been changed from the original version.
Overwrite:
[x] Exit
[y] Yes
[n] No
[d] Diff

y
/etc/asterisk/cdr_adaptive_odbc.conf has been changed from the original version.
Overwrite:
[x] Exit
[y] Yes
[n] No
[d] Diff
y
6165/6165 [============================] 100%
Done
Finishing up directory processes...Done!
Creating missing #include files...Done
Running variable replacement...Done
Setting up Asterisk Manager Connection...Done
Running through upgrades...
Checking for upgrades..
No further upgrades necessary
Finished upgrades
Setting FreePBX version to 13.0.53...Done
Writing out /etc/amportal.conf...Done
Setting Permissions...
58/58 [============================] 100%
Finished setting permissions
Installing all modules...checking for sipsettings table..already exists
Generating CSS...Done
Module sipsettings successfully installed
Updating Hooks...Done
Generating CSS...Done
Module music successfully installed
Updating Hooks...Done
Generating CSS...Done
Module callrecording successfully installed
Updating Hooks...Done
Creating cel if needed..OK
checking for extra field..already exists
checking for userfield field..already deleted
checking for context index..already indexed
Generating CSS...Done
Module cel successfully installed
Updating Hooks...Done
Generating CSS...Done
Module dashboard successfully installed
Updating Hooks...Done
Generating CSS...Done
Module featurecodeadmin successfully installed
Updating Hooks...Done
Generating CSS...Done
Module infoservices successfully installed
Updating Hooks...Done
Generating CSS...Done
Module userman successfully installed
Updating Hooks...Done
Checking if field did is present in cdr table..
did field already present.
Checking if field recordingfile is present in cdr table..
recordingfile field already present.
Checking if field cnum is present in cdr table..
cnum field already present.
Checking if field cnam is present in cdr table..
cnam field already present.
Checking if field outbound_cnum is present in cdr table..
outbound_cnum field already present.
Checking if field outbound_cnam is present in cdr table..
outbound_cnam field already present.
Checking if field dst_cnam is present in cdr table..
dst_cnam field already present.
Generating CSS...Done
Module cdr successfully installed
Updating Hooks...Done
Checking if directdids need migrating..already done
updating zap callgroup, pickupgroup..not needed
checking for delay_answer field ..already exists
checking for reversal field ..already exists
checking for pricid field ..already exists
Checking if trunk table migration required..not needed
Checking if privacy manager options exists..already exists
Checking for noanswer_cid field..already exists
Checking for busy_cid field..already exists
Checking for chanunavail_cid field..already exists
Checking for noanswer_dest field..already exists
Checking for busy_dest field..already exists
Checking for chanunavail_dest field..already exists
Unable to add index to extensions field in users
Checking for General Setting migrations..not needed
Deleting unused globals..done
Converting IAX notransfer to transfer if needed..updated 0000 records
deleting obsoleted record_in and record_out entries..ok
checking for dest field in outbound_routes..already exists
checking for continue field in trunks..already exists
upgrading any zap trunks to dahdi if found..ok
checking possibly for invalid emergency caller id fieldsNo invalid callerid entries foundGenerating CSS...Done
Module core successfully installed
Updating Hooks...Done
Generating CSS...Done
Module logfiles successfully installed
Updating Hooks...Done
Checking for General Setting migrations..not needed
checking if Voicemail Admin (vmailadmin) is installed..not installed, ok
Generating CSS...Done
Module voicemail successfully installed
Updating Hooks...Done
Refreshing all UCP Assets, this could take a while...
Generating Module Scripts...Done
Generating Module CSS...Done
Generating Main Scripts...Done
Generating Main CSS...Done
Done!
Generating CSS...Done
Module ucp successfully installed
Updating Hooks...Done
Generating CSS...Done
Module customappsreg successfully installed
Updating Hooks...Done
Done installing modules
Installing framework...
No directory /var/www/html/admin/modules/framework/amp_conf/htdocs, install script not needed
Generating CSS...Done
Module framework successfully installed
Updating Hooks...Done
Done
Generating default configurations...
PHP Fatal error: Class 'FreePBX\modules\Userman\Auth\freepbx' not found in /var/www/html/admin/modules/userman/Userman.class.php on line 991
Whoops\Exception\ErrorException: Class 'FreePBX\modules\Userman\Auth\freepbx' not found in file /var/www/html/admin/modules/userman/Userman.class.php on line 991
Stack trace:
1. () /var/www/html/admin/modules/userman/Userman.class.php:991
Finished generating default configurations
Trusting FreePBX...Trusted
Setting Permissions...
58/58 [============================] 100%
Finished setting permissions
You have successfully installed FreePBX
`

installed q xact reports and am trying to put together a report that shows all queue activity since the beginning of the year. there are four queues, but only two show up in the report option. there are also around 20 agents but only 6 show up in the report. running a report for month (jan 1- jan 27) gives the same data as running the report for today.
the system is running PBX Firmware: 6.12.65-31 i cant figure out how to get the other queues or agents added to the report. isymphony shows that one queue has had over 400 calls, but qxact, regardless of the time internal shows only 6

unless i am misunderstanding your question, you can do the same thing on the voip phone as you are doing on the traditional pbx, namely forward the calls to a cell phone. they can do this manually on the phone or use UCP to do it via a web browser.

Hello,

You mentioned that the ifcfg-eth0 settings are correct. Let's start there.

Please boot up your FreePBX box, and login to the text based console. You might see an ASCII art text box with your interface, MAC addresses, and IP information.

This is what mine looks like:

Current Network Configuration
+-----------+-------------------+--------------------------+
| Interface | MAC Address       | IP Addresses             |
+-----------+-------------------+--------------------------+
| eth0      | 00:90:27:ED:69:2A | 192.168.1.50             |
|           |                   | fe80::290:27ff:feed:692a |
| eth1      | 00:90:27:ED:69:2B |                          |
| eth2      | 00:90:27:ED:69:2C |                          |
+-----------+-------------------+--------------------------+

You should then type in

route -n

and get an answer similar to:

root@pbx ~]# route -n
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
192.168.1.0     0.0.0.0         255.255.255.0   U     0      0        0 eth0
169.254.0.0     0.0.0.0         255.255.0.0     U     1002   0        0 eth0
0.0.0.0         192.168.1.1     0.0.0.0         UG    0      0        0 eth0

Note the entry of 0.0.0.0 as this is your default gateway, in my case 192.168.1.1

Try to ping your default gateway. Also try to ping another machine on that subnet. In my case, I would try 192.168.1.1 and another device, such as a phone, at 192.168.1.200 and your setup will certainly be different.

Note that your router can have a firewall that declines pings, therefore having another device on the same subnet is handy for testing.

My gut guess is that you have:
-- No default gateway defined
-- Bad subnet mask
-- IP conflict

You can also go to your router, and see if the router sees (pings) your FreePBX machine.

If you want to get really fancy, you can try tcpdump in a second session, and perform a packet sniff, or you can put a hub (or a port mirror on a switch) and use something like wireshark to see what the network is doing.

Christian

In addition to @cyberdocwi's awesome debugging instructions, I would also check that the network interface you THINK is eth0 is actually eth0.

Try using the command 'mii-tool eth0' and making sure it is actually reporting up and connected when you're expecting it to be.

You may need to move a network cable to a different port.

FreePBX12 distro installation. Overall the system works. The problem is that both inbound and outbound calls have a 10 to 15 second delay between dialing the number and the first ring.

The problem appears to be some kind of configuration problem on my system. For example, when calling in from outside, tail -f /var/log/asterisk/full shows that the call is routed from Vitelity to my system within a second or two. The log spins through to this point, which shows a 9 second delay.

> [2016-01-25 10:40:07] VERBOSE[11808][C-00000073] pbx.c: -- Executing [s@app-blacklist-check:3] Return("SIP/vitel-inbound-00000076", "") in new stack
> [2016-01-25 10:40:07] VERBOSE[11808][C-00000073] pbx.c: -- Executing [s@ext-did:3] Gosub("SIP/vitel-inbound-00000076", "cidlookup,cidlookup_2,1()") in new stack
> [2016-01-25 10:40:07] VERBOSE[11808][C-00000073] pbx.c: -- Executing [cidlookup_2@cidlookup:1] Set("SIP/vitel-inbound-00000076", "CURLOPT(httptimeout)=7") in new stack
> **[2016-01-25 10:40:08] VERBOSE[11808][C-00000073] pbx.c: -- Executing [cidlookup_2@cidlookup:2] Set("SIP/vitel-inbound-00000076", "CALLERID(name)=EVOLVING SYS") in new stack**
> **[2016-01-25 10:40:23] VERBOSE[1788][C-00000074] netsock2.c: == Using SIP RTP TOS bits 184**
> [2016-01-25 10:40:23] VERBOSE[1788][C-00000074] netsock2.c: == Using SIP RTP CoS mark 5
> [2016-01-25 10:40:23] VERBOSE[11854][C-00000074] pbx.c: -- Executing [6503252100@from-trunk:1] NoOp("SIP/vitel-inbound-00000077", "Catch-All DID Match - Found 6503252100 - You probably want a DID for this.") in new stack
> [2016-01-25 10:40:23] VERBOSE[11854][C-00000074] pbx.c: -- Executing [6503252100@from-trunk:2] Log("SIP/vitel-inbound-00000077", "WARNING,Friendly Scanner from 66.241.99.27") in new stack
> [2016-01-25 10:40:23] WARNING[11854][C-00000074] Ext. 6503252100: Friendly Scanner from 66.241.99.27
> [2016-01-25 10:40:23] VERBOSE[11854][C-00000074] pbx.c: -- Executing [6503252100@from-trunk:3] Set("SIP/vitel-inbound-00000077", "__FROM_DID=6503252100") in new stack

Need a bit of help identifying the configuration problem and how to correct it.

Thanks.

Freepbx 13.0.51

ucprecord.png800x258 111 KB

recording option.png983x311 23.2 KB

I have several ext where the recording option is set the same (see pic). Out of 7 ext, one which is connected to Cisco 112 (ATA) started recording all conversations. I can set it to no or never but wondering what triggers it to start recording.

I like the feature and might have a use for it in the future but are there any setting how much history (or just recording) to keep track. Is there a way to delete the recording. I can download but that all the option I can see.. Just want to minimized the disk space usage.

Thanks

What version of PHP is this using?

In the module call recording reports which is commercial.

Otherwise I suggest you open a feature request for this.

How do I remove all agents from a queue?

jason