This is still an issue in FOP2 Version: 2.31.23.

My above workaround still works, but it would be nice to see a fix coded into the program itself.

So, I noticed that there was a new version of FOP2, ver 2.31.21 that came out 17 December 2018. I have installed that and it seems to have fixed my issues.

I haven't yet had a chance to give the system a full live workout, but it is now handling scenarios that were consistently breaking it on 2.31.20 - so I'm feeling optimistic.

Hi Nicolas.
New installation of FOP2, ver 2.31.20, installation is on Centos 7 using Freepbx 14.0.5.1, asterisk version 13.23.1
i used the 'automagic' installation method: wget -O - http://download.fop2.com/install_fop2.sh | bash

All seems working except that the conferences (I am using the Freepbx Conference module) do not show who is in the conference, and do not detect when the conference is over. The button shows 'in use' colour, but no members.

When everyone hangs up, the button stays lit. I have to restart fop2 to clear the 'in use' colour on the button.

Can you assist? Thanks in advance.

I took a long time to get around to this, but I have had a play, and I think I know what is going on. I have set up a workaround, but I also think I know what is needed for a proper fix.

THE PROBLEM
When a user dialled from an extension (3 digits) the lookup would often incorrectly return a name and picture associated with a different address book entry. This happened when the 3 digit extension number appeared anywhere within the longer phone number. For example, calling extension 650, would select entry for user with phone number 12346501

The issue is that the logic of this select statement did not work:

$res = $db->consulta("SELECT concat(firstname,' ',lastname) AS name,company,picture FROM visual_phonebook WHERE (phone1 LIKE '%%%s'OR phone2 LIKE '%%%s') AND context='%s' ORDER BY LENGTH(CONCAT(phone1,phone2)) LIMIT 1",$clid_significant,$clid_significant,$context);

The problem is that the LIKE statement was finding all records containing the callerid phone number ANYWHERE within it. So, when the calling extension is 650, it matched on ALL of 1236504, 1234650, 650, 16501245. The ORDER BY statement was then supposed to select the shortest match (which should have been the extension). However, because there are two phone fields, an attempt was made to identify the record with the shortest match by concatenating both numbers in the record, and comparing that length with the concatenated numbers of the other matching records.
The LIMIT 1 statement then picked the first record.

This does not work because there is no certainty that the total length of both phone number fields will be shorter for the correct entry.

Consider this mini-database:
Calling Extension is 650.
User A Phone1: 650 Phone2: 12345678
User B Phone1: 12650348 Phone2: nul

Both of these records are selected by the above statement. They are then compared by adding the two phone fields, and picking the shortest COMBINED length:
User A: 3+8=11
User B: 8+0=8

User B record is shorter, so it is selected. The correct record is User A.

MY WORKAROUND
In Australia, all phone numbers are 10 digits (including area code) – even mobile numbers.
So, I have set the comparison to EQUALS (rather than LIKE) so that I only get one valid match. I have removed the LENGTH comparison altogether.

$res = $db->consulta("SELECT concat(firstname,' ',lastname) AS name,company,picture FROM visual_phonebook WHERE (phone1 = '%s' OR phone2 = '%s') AND context='%s' LIMIT 1",$clid_significant,$clid_significant,$context);

This works, except that if the visual phone book entry does not include the area code, or includes spaces, the record is not found.

FIX REQUIRED
• We need to be able to strip the spaces out of the visual phonebook phone number entries when comparing;
• We need to be able to perform a routine on the visual phonebook phone number entries to only compare the significant digits – in the same way that the script does with the caller id number:

	$significant    = 10;
	$startoffset    = 0;
	if($largo > $significant) {
        $startoffset=$largo-$significant;
    }

    $clid_significant = substr( $decodedClidnum, $startoffset );

When comparing the CLID with the visual phone book entries after stripping spaces and looking at the same significant digits, the EQUALS comparison operator will then work as desired.

I am not sufficiently expert to figure out how to do this bit. Hopefully what I have said helps, and the proper fix will be simple for you

Hi Nikolas,
I found this thread because I am experiencing similar issues.

Forgot to add, you can also add your own extensions in the phonebook, so they will match instead of the other customers. If you have an entry with a phone 52 it will have precedence over 3928237352

This does not appear to be the case. My extensions are 3 digits long. When there is an address book entry for that extension, say, 215, and there is another address book entry for someone else , say 55334215, the second (longer) entry is matched, not the shorter (more precisely matching) entry.

Running FOP2 version 2.31.05 on FreePBX 13.0.191.11. Any thoughts?

I am confused now too.

I have my setup so that, in the FOP2 User setup, the "Secret" field is left blank. The information popup for this option says: "The secret to login to fop2. If none specified the voicemail secret will be used."

I have tested this, and if I change my voicemail password, the NEW password is what I have to use to log into FOP2. That is, I use my current voicemail password, whatever it is.

My only caveat is that I am running FOP2 2.2.7. However, this has been the behaviour since I first started using FOP2, which was a few years and a few versions earlier still.

Hi.

Do I need to change any configs to allow the use of FOP2 connectivity using IPV6?

I am running Freepbx 2.11.0.38 on a CentOS 64 vm. I am running FOP2 v 2.2.7

I have recently enabled IPv6 on the asterisk server (dual stack), and can browse the Freepbx asterisk web interface via IPv6.

Any attempts to connect to FOP2 time out. I can log into the FOP2 server via ipv4 as normal.

In order to IPV6 enable the SIP server, I had to change the asterisk HTTP bind address from "0.0.0.0" (any IPv4 address) to "::" (any IPv4 or any IPv6 address).

Looking at netstat, I think FOP2 is still only listening on IPv4 addresses:

netstat -lnp | grep 4445
tcp        0      0 0.0.0.0:4445                0.0.0.0:*                   LIST                   EN      30108/fop2_server

Does FOP2 allow connection via IPv6? How do I enable it?

A re-install with the latest version seems to have fixed all issues.

Thanks for your help.

A re-install with the latest version seems to have fixed all issues (including my recording issues). I must've just been unlucky to have downloaded the new 2.27 version too early.

I really like the plug-in setup - great for control and visibility of options.

Thanks for your help.

One tiny point, for your next fix - the drop down box of permissions 'Quick Pick' does not include the 'presencemanager' permission - it has to be manually typed in.

The PresenceDND plugin in FOP 2.27 was updated to support BLF, you should update the plugin

sorry, how do I do that? :oops:

While I said that I 'upgraded' to FOP2 2.27 in my post, it would be more accurate to say that I did a fresh install of FOP2 on a fresh install of FreePBX/Asterisk/Centos 6.5 (64 bit), and then re-created my setup on the new system, then stopped using my old system.... ie - as far as the machine I am using now is concerned, it was a fresh install. Only the user browsers will have 'old' content, and I have Ctrl-R as hard as I can go with all affected browsers before reporting the issues.

I note that I can do the 'old way'.

Or you can run the script generate_override_context.pl -w --force to generate the old style of dialplan configs for FreePBX.

I would prefer to go with the 'new way' if I can get it working - to maximise compatibility with any future work you do on the panel.

Lastly, while it all appears to be working well, I do get some errors when I run --test

[root@pbx fop2]# ./fop2_server --test
unable to initialize libusb: -99
Flash Operator Panel 2 - Valid License (7)

/usr/local/fop2/autoconfig-buttons-freepbx.sh: line 17: [: too many arguments
Connection to manager OK!
[root@pbx fop2]#

I did the initial install probably 2 months ago - have there been changes to the package (with same build number) since that time?

Not sure if this is relevant, but I note I am having issues with recordings as well :
[url:1urjqkyg]http://forum.fop2.com/esotalk/index.php/1160&start=10[/url:1urjqkyg]

so is there something I can/should do to re-fresh my install?

Thanks for your help.

It looks like recordings.php is already set to allow persons with 'phonebook' permissions to view recordings:

$context   = $_SESSION[MYAP]['context'];
$extension = $_SESSION[MYAP]['extension'];
$allowed   = $_SESSION[MYAP]['phonebook'];
$admin     = $_SESSION[MYAP]['admin'];

if($allowed <> "yes") {
   die("You do not have permissions to access this resource.");
}

and admin or ownerextension to play them:

$grid->set_condition("(ownerextension='$extension' OR $admin=1)");

However this is not the actual behaviour I am seeing. What is happening is that if the user has anything other than 'all' permission set, then they cannot see anything at all under the "Recordings" tab. If they have the 'all' permissions set, then they see ALL recordings, made by any extension.

Am I right in thinking that this is not the designed behaviour? According to the script (which I have not modified) shouldn't a user with 'phonebook' permission already be able to see their own list, and play it ? If so, any theories as to what is going wrong? (Or am I misreading the script entirely :oops: )

Hi Nicolas,

I have recently upgraded to a new Freepbx system, and the FOP2 ver 2.27
I am running:
FOP2 version 2.27
FreePBX 2.11.0.24
CentOS 6.5

I am using Grandstream GXP2120 handsets, with BLFs set to monitor the state of various extensions.

Mostly everything is working well, but an old problem [url:2hf2vro9]http://forum.fop2.com/esotalk/index.php/1282[/url:2hf2vro9] has resurfaced.

The problem is that if I set a DND via the FOP2 panel, this does not change the BLF to red on the phone buttons monitoring that extension.

The reverse DOES work - If I set the phone to DND by dialling *76, then any BLF monitoring that extension turns red, and the DND indicator appears on the FOP2 panel. Using *76 on the phone correctly toggles the status, and the BLF and FOP2 panels correctly update.

Now, last time this issue arose (as per old thread linked above) the solution was to edit the FOP2Callbacks.pm file.

When I view this file now, in version 2.27, it says

DND Integration from FOP2 to FreePBX ASTDB DND

    #
    # DEPRECATED! All of this can be done from the new plugin
    # architecture

So, what do I got to plug in where to make my pbx recognise DND settings made from the FOP2 panel ? :?

Thanks in advance.

The remaining issue is that my 'normal' fop2 users cannot see or play any recordings unless they have the 'all' permission.

I have tried giving the users pretty much every permission individually, and in every case, they cannot see nor play the recordings. If I give them the 'all' permission, they can see and play the recordings.

Specific permissions I have tested, none of which allow viewing or playing recordings are:
chat
dial
hangupself
meetme
phonebook
pickup
preferences
queueagent
queuemanager
record
recordself
transfer
transferexternal

I would prefer not to give the 'all' permission to all of my users. Is there a better solution?

Hi Nicolás,
Digging up an old post here, but it seems that the issue identified in this post is not wholly resolved

I found this post by googling this site for the same issue - namely, a 'no way' message when a user tried to play or download a recording.

I am using:
FOP 2.27
Asterisk 11.7.0
FreePBX 2.11.0.24
Running on CentOS 6.5

I see that, in FOP2.27 you have incorporated the coding change mentioned in this post into the download.php file
(for the benefit of other readers, this file is located here: /var/www/html/fop2/download.php )

Even with this change, I still experienced the issue. I have looked at my config, and the FOP2 recordings are no longer being stored in /var/spool/asterisk/monitor/fop2 ;
instead, they are just being stored in /var/spool/asterisk/monitor

accordingly, I changed this line in download.php:

if(!stristr($_SESSION[MYAP]['vfile'],$_SESSION[MYAP]['vpath']) && !stristr($_SESSION[MYAP]['vfile'],'/var/spool/asterisk/monitor/fop2')) {

to remove the 'fop2', becoming:

if(!stristr($_SESSION[MYAP]['vfile'],$_SESSION[MYAP]['vpath']) && !stristr($_SESSION[MYAP]['vfile'],'/var/spool/asterisk/monitor')) {

This now works. 8-)

There is a sub-issue remaining, which I will post in a separate post below.

Ok, just to test, I changed it back to the IP address, and... it continued to work.
so, I don't know what it was !?!

All I can say for sure is: when I used the IP address, it refused to accept the credentials. When I used DNS name, it worked - no other changes. Now, it seems to be happy either way.

Hi
I have the FOP2 Lite edition working as a trial and can login and access it using Chrome on my PC. This is working perfectly and pops open the caller id information etc.

I have now added the Chrome extension to get the Click to Dial option working, however using the same credentials as I use for the FOP2 desktop the extension just says that it is unable to connect. The only parameters I have included are:

• the FOP2 Server
• Port I have left at the default
• Extension - same as I use to login to FOP2
• Password same as I use to login to FOP2
• Context - left bank

Any ideas on what the issue might be or how to troubleshoot the connection issue?

I found this thread when experiencing the same problem. (The thread is highjacked , and the original question was not resolved).

I was able to resolve this problem by trial and error. I thought I would post up what I found, as it was not obvious (to me), and not documented.

Solution:
In the "Fop2 Server" dialog box, enter the address of the server as a dns name, not as an IP address, and it works.

If the IP address is entered, it indicates: invalid credentials.
changing it to a dnsname, such as: abc.yourdomain.local with no other changes - works immediately.

WFM. Hope this helps others.

Ok, thanks Nicolas.

I will keep looking, and post back if/when I locate/resolve the underlying problem!

Thanks for your further input - I have it working now!

This from your last post was the key:

You can look at the recordings_fop2.pl script

I checked this file, and the comments at the top, and found that my permissions on this file were set to 'root' and not the 'asterisk' account I am using for asterisk.
It also pointed me to the fop2.cfg file, where I found this:

; To enable the recording interface you must uncomment the following
; line, but also you might need to modify the script a little bit 
; depending on the sox version you have installed.
;
;monitor_exec=/usr/local/fop2/recording_fop2.pl

So, I uncommented the line

monitor_exec=/usr/local/fop2/recording_fop2.pl

Then, recordings started to work, mixed beautifully. The final recording is in stereo - with the playback playing my caller sound on the left channel, and my callee on the right channel, just like a conversation. Wonderful!

It is now saving the completed recording using path /var/spool/asterisk/monitor/fop2/[date] rather than just leaving it in the top level /var/spool/asterisk/monitor/

I can now 'see' the recording in my phonebook.

I didn't even have to restart fop2.

All working, 100% A OK

Now, just a couple of other matters, for anyone else finding this topic:
My issue waswith FOP2, in that I did not have FOP2 config set up correctly.
I did not know I had to uncomment lines in a config file to enable recording.

I think the 'not putting the recorded file into a subdirectory using date' and 'recording not visible in phonebook' were, in hindsight, the key indicators of [my] user config error.

A couple more things, more for completeness, and because I tried a lot of stuff so may as well note it down:

Also in the fop2.cfg file is this section

; You could specify your own script to be executed when the recording
; is finished. It will receive 3 parameters, the complete
; path and filename of the IN leg, the OUT leg and the final
; recording NAME. You should run soxmix in your script to join
; the recordings into one file.
;
; monitor_exec=/var/lib/asterisk/bin/postrecording-script.sh

I did not have to uncomment the postrecording-script.sh line, nor specify my own script to be executed when recording. This was to be my next step, but it was not required.

Also:
I do not have a file called 'res_monitor.c' on my system.

I tried mixing the in and out channels of a test recording using both of these suggestions:

nice -n 19 soxmix wwwww-in.wav xxxxxx-out.wav xxxxxx-combined.wav

or

nice -n 19 sox -m wwwww-in.wav xxxxxx-out.wav xxxxxx-combined.wav

One concatenated the file, the other threw an error, and created a file with only the 'in' channel. So, this did not work.

I was very reluctant to change my sox version as, as pointed out by Nicolas, asterisk 'expects' a certain version and my *1 recording method was working and I didn't want to break it. Part of my research suggested to me that asterisk is now using its own recording method, not using sox at all, but I didn't really get to the bottom of that.

Thanks again Nicolas for an excellent program. My users are all hugely resistent to any form of technology, particularly remembering and dialling codes into the phone, so your big colourful buttons and point-and-click panel is very popular. My questions and queries are not a complaint, they are the product of a genuine desire to get the best out of a very good system.

HI Nicholas,
I am sorry I have taken such a long time to reply. I have been looking into this, and running some tests.

I have read, very carefully, the post you linked to me about the sox version.

I note that upgrading the sox version did not fix the issues being experienced. I have tried this 'fix' which worked for some of the users on the post, but did NOT fix my issue:

Try putting the following line in /etc/asterisk/globals_custom.conf:
MONITOR_EXEC=soxmix

When I did this, the only effect was that the FOP2 recordings stayed separate (ie, two recordings, one 'in' and one 'out') when without that line in the globals_custom.conf the two files are appended into one file.

Dialling *1 in call works perfectly for me - whether the asterisk system is the callee, or the caller.

It seems to me that, actually, the problem really is with FOP2 .

The command that FOP2 issues to make the recording must be different to the command issued when I dial *1 in call.

When the 'record' option is used in FOP2, the recording is placed directly into /var/spool/asterisk/monitor, while when done via *1, the recording is placed in /var/spool/asterisk/monitor/year/month/day

Also, the FOP2 phonebook "Recordings" tab shows "No records found" even though it just made a recording.

I have had a look and FreePBX is using the following line in /etc/asterisk/features_applicationmap_additonal.conf

apprecord=>*1,caller,Macro,one-touch-record

I am not sure where to look to find the macro itself..

It seems this macro works fine, but the fop2 one doesn't? Is it the case that the fop2 command is issued expecting a different sox version than exists on my system, while the freepbx ver is using a command compatible with my sox ver? I have sox ver 12.18.1, which I see is quite old, yet seems to be the latest version available in the centos repository my system is using.

Any suggestions where to go from here?

Thanks in advance.

Hi all, I have a FreePBX 2.10.1.1 install, on Cent0S 64bit

Linux version 2.6.18-274.3.1.el5 (<!-- e --><a href="mailto:mockbuild@builder10.centos.org">mockbuild@builder10.centos.org</a><!-- e -->) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-51)) #1 SMP Tue Sep 6 20:13:52 EDT 2011

My basic problem is that recordings made using the FOP2 interface are recording the 'in' and 'out' channel separately, and then appending them into a single .wav file - rather than mixing them so that you can hear both sides of the conversation at the same time.

So, I end up with one recording, twice as long as the call took, the first half of the recording containing anything said by the caller, and the second half being anything said by the callee. This is rather difficult to interpret!

This does not happen when I record the call using *1 in call. In that case I get, correctly, a single .wav file with both sides of the conversation.

I also note that the bitrate of the FOP2 recording is much higher.

Is this a feature or a bug? Is there some way I can get the FOP2 recordings to mix the 'in' and 'out' so that I can hear the whole conversation?

I had a bit of a look to try and find the FOP2 recording settings, with no success. Somewhere there must be the ability to set the bitrate, and, probably the recording format?

Thanks for any help. I apologise in advance for what may turn out to be a dumb question

PS: Also, FOP2 seems unaware of the recording in progress when the recording is vial *1. That is, when I record using FOP2, I get a cute little tape icon on the button for the call being recorded. When I use *1, the button does not indicate that the call is being recorded. Not a big issue, but perhaps a symptom?