Good day ,
i installed the new version ASTERISK NOW 3.0.0 from cd. I burned the cd and everything was ok. After installation i made yum update -y and after this 2 problems appear:
1. any time i move from one page to another the Login is needed to enter the page.
2. If i enter in the Module Admin page and try to make updates the appear the issue :

Error: Did not receive valid response from server

XHR response code: 0 XHR responseText: undefined jQuery status: error

Can somebody solve this problem? Thanks in advance !!

Hi all.

I'm new bie. recently I install sip trunking-VTN_Inbound- (asterisk 1.8.x x64) with VTN, but when "sip show peers" the sip trunking status is unreachable. we captured log on VTN, and we got "401 unauthorized error". I sent my sip trunking configruation here:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

[223_In]
host=192.168.222.223
port=5060
type=peer
qualify=yes
context=from-trunk

[223_Out]
host=192.168.222.223
port=5060
type=peer
qualify=yes
context=from-trunk-sip-223_Out

[6001]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6001
mailbox=6001@device
permit=0.0.0.0/0.0.0.0
callerid=device <6001>
callcounter=yes
faxdetect=no

[6002]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6002
mailbox=6002@device
permit=0.0.0.0/0.0.0.0
callerid=device <6002>
callcounter=yes
faxdetect=no

[6003]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6003
mailbox=6003@device
permit=0.0.0.0/0.0.0.0
callerid=device <6003>
callcounter=yes
faxdetect=no

[6004]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6004
mailbox=6004@device
permit=0.0.0.0/0.0.0.0
callerid=device <6004>
callcounter=yes
faxdetect=no

[6005]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6005
mailbox=6005@device
permit=0.0.0.0/0.0.0.0
callerid=device <6005>
callcounter=yes
faxdetect=no

[6006]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6006
mailbox=6006@device
permit=0.0.0.0/0.0.0.0
callerid=device <6006>
callcounter=yes
faxdetect=no

[6007]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6007
mailbox=6007@device
permit=0.0.0.0/0.0.0.0
callerid=device <6007>
callcounter=yes
faxdetect=no

[6008]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6008
mailbox=6008@device
permit=0.0.0.0/0.0.0.0
callerid=device <6008>
callcounter=yes
faxdetect=no

[6009]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6009
mailbox=6009@device
permit=0.0.0.0/0.0.0.0
callerid=device <6009>
callcounter=yes
faxdetect=no

[6011]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6011
mailbox=6011@device
permit=0.0.0.0/0.0.0.0
callerid=device <6011>
callcounter=yes
faxdetect=no

[6012]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6012
mailbox=6012@device
permit=0.0.0.0/0.0.0.0
callerid=device <6012>
callcounter=yes
faxdetect=no

[6013]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6013
mailbox=6013@device
permit=0.0.0.0/0.0.0.0
callerid=device <6013>
callcounter=yes
faxdetect=no

[6014]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6014
mailbox=6014@device
permit=0.0.0.0/0.0.0.0
callerid=device <6014>
callcounter=yes
faxdetect=no

[6015]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6015
mailbox=6015@device
permit=0.0.0.0/0.0.0.0
callerid=device <6015>
callcounter=yes
faxdetect=no

[6016]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6016
mailbox=6016@device
permit=0.0.0.0/0.0.0.0
callerid=device <6016>
callcounter=yes
faxdetect=no

[6017]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6017
mailbox=6017@device
permit=0.0.0.0/0.0.0.0
callerid=device <6017>
callcounter=yes
faxdetect=no

[6018]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6018
mailbox=6018@device
permit=0.0.0.0/0.0.0.0
callerid=device <6018>
callcounter=yes
faxdetect=no

[6019]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6019
mailbox=6019@device
permit=0.0.0.0/0.0.0.0
callerid=device <6019>
callcounter=yes
faxdetect=no

[6021]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6021
mailbox=6021@device
permit=0.0.0.0/0.0.0.0
callerid=device <6021>
callcounter=yes
faxdetect=no

[6022]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6022
mailbox=6022@device
permit=0.0.0.0/0.0.0.0
callerid=device <6022>
callcounter=yes
faxdetect=no

[6023]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6023
mailbox=6023@device
permit=0.0.0.0/0.0.0.0
callerid=device <6023>
callcounter=yes
faxdetect=no

[6024]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6024
mailbox=6024@device
permit=0.0.0.0/0.0.0.0
callerid=device <6024>
callcounter=yes
faxdetect=no

[6025]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6025
mailbox=6025@device
permit=0.0.0.0/0.0.0.0
callerid=device <6025>
callcounter=yes
faxdetect=no

[6026]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6026
mailbox=6026@device
permit=0.0.0.0/0.0.0.0
callerid=device <6026>
callcounter=yes
faxdetect=no

[6027]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6027
mailbox=6027@device
permit=0.0.0.0/0.0.0.0
callerid=device <6027>
callcounter=yes
faxdetect=no

[6028]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6028
mailbox=6028@device
permit=0.0.0.0/0.0.0.0
callerid=device <6028>
callcounter=yes
faxdetect=no

[6029]
deny=0.0.0.0/0.0.0.0
secret=ab123456
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/6029
mailbox=6029@device
permit=0.0.0.0/0.0.0.0
callerid=device <6029>
callcounter=yes
faxdetect=no

[VTN_Inbound]
host=10.163.1.57
port=5060
type=friend
canreinvite=no
context=from-trunk
qualify=yes
allow=all
alwaysauthreject=no

[VTN_Outbound]
host=10.163.1.57
port=5060
type=peer
type=friend
context=from-trunk
qualify=yes

So can any pro brother give my a issue to solve it?

Thank all.

Sorry in advance for the long thread. I'm new to FreePBX and would like to use AJAM. Is it is possible, and if so how do you install it?

So far my searching has turned up more questions than answers For example:

1. Can AJAM be enabled/installed separately?
   Searching turned up this post about PBX In a Flash (which includes FreePBX)

   http://nerdvittles.com/?p=234

   which says it "comes with AJAM support built in but not activated". That suggests it can be installed separately, but I'm not clear on how/if it relates to my FreePBX distro.

2. Is Noojee Click required?
   I also found this post which mentions Noojee Click

   http://www.freepbx.org/forum/freepbx/users/ajam-aynchronous-javascript-asterisk-manager-how-to-set-it-up

   ... but that sounds like a separate module (for lack of a better word) on top of AJAM. Is Noojee Click required to get AJAM working w/FreePBX?

3. What are the implications of using AJAM w/FreePBX
   Also, the end of that thread mentions being mindful of the implications. Can anyone elaborate on what that means?

   ".. FreePBX since it does not support the AJAM directly
   you must be mindful of the implications of running
   commands on a system with FreePBX vs. a bare metal
   Asterisk system...."

Any insights - to even part of my questions - would be greatly appreciated

---------------
Note: I'm using the latest distro from Schmooze
Stable-4.211.64-9
Release Date-12-01-13
FreePBX 2.11, Centos 6.4
Asterisk 11

Hi,

I have purchased a Digium TE110 E1 PRI PCI based card.
Installed it in a core i5 System and then installed FreePBX
I now have E1 PRI line from BSNL, India.

After going to the other tab of the FreePBX menu, I get the PRI card in recovery mode.
I have configured it as follows:
Alarms:REC
Framing/Coding:
Channels:31/31 (E1)
Signalling:PRI-CPE
Switchtype:National ISDN
Sync/Clock Source:0
Line Build Out:0
Pridialplan:UnKnown
Prilocaldialplan:UnKnown
Group: 0
Context: incoming
Channels:31
From: 1-15,17-31 Reserved: 16

Now I have to make sure that I would be able to do incoming calls.
Do I need to do any further steps than this?

Regards,
Piyush

If anyone knows how to setup FreePBX up on a VPS for a price feel free to contact me on skype - yaz_1994.

Regards

Greeting Forum,

First time user here, so please be patient if I describe things wrong... I have searched around but can not seem to find an answer to my issue, so I am posting it as a new topic.

I just recently downloaded and installed the FreePBX Distro, as we are leaving our hosted provider and bringing everything in-house. The install went smooth, and all our phones are working, thanks to purchasing the commercial Endpoint Manager from Schmoozecom.com, but there is one nagging issue...

We have a second asterisk device, a predictive dialer, and I am having a hell of a time linking the two together. Currently we are using a SIP trunk on the FreePBX with an outbound route and dial patterns to connect those that need access. This works fine, however after getting about 20 people connected, QoS nosedives to the point that we only hear every other word the person on the other line is saying.

Each device has 2 NICs, with eth0 on both going to my local network, and Eth1 going to a private Voice circuit, completely segmented from the internal network.

Using this setup, a regular telephone call looks like this:

Phone -> Switch -> Core Switch -> PBX Eth0 -> PBX Eth1 -> Cisco Voice Switch -> Provider

And a Dialer Call looks like this:

Phone -> Switches -> Core Switch -> PBX Eth0 -> Core Switch -> Dialer Eth0 -> Cisco Voice Switch -> Provider

As you can see, there is an extra hop back to the Core switch on dialer calls. Multiple that by the 75 agents, and 30 admin people making calls and using data...

Unfortunately, at this time, we are not using VLans, but that is being fixed next week. But the hardware is all higher end HP/Cisco, so it isn't like I am using netgear or linksys.

If there is no "fix" then I would have to provision the phones to have a direct connection to the dialer, as well as the PBX, but from what I can see in Endpoint Manager, that isn't doable.

Any insight you guys can give, would be great!

Brian Carson
IT Manager

Hi,

My Incoming Numbers not showing for incoming calls, I have setup ZAP CHANNEL DIDs Port Numbers.

When I place inbound call route to IVR, i get message "the number you are calling is not in service", but when i add ANY DID /ANY CLID, I get the call successfully. I checked the CALLER ID and its works fine from ISP, but sometime it shows the number and sometime it doesnt show UNKOWN NUMBER.

Is there any security blocking to view incoming numbers, Because when I connect an analog phone direct to my landline, i can able to see the numbers. But when i connect the same landline to my asterisk, then i can able to see the incoming numbers sometime and sometime UNKNOWN NUMBER.

May I know how do i correctly configure the DID Numbers route to my IVR.
Is there any configuration to avoid blocking to view incoming numbers ?
Is there any extra configuration require or diagnose the problem.

Your repsonse will be highly appreciated

Thanks,
Sam.

I am installing freePBX in the company i work for.
It's rather large place, and there's a lot of NAT'ing and the topology is somewhat unclear for me, and i'm struggling with the admins about what's right and what's not.

Here is what I know:

FreePBX box has internal ip and external IP (see the log for more info)
I can make SIPcalls from my phone using 3G network to the PBX with sound.
I can see that it's ringing when i try to call the phone, but i cannot take it.

I can see that i get this warning in the log:
[2013-12-06 10:07:40] WARNING[2201] chan_sip.c: Retransmission timeout reached on transmission

:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

I've turned on SIP Debug, and made a testcall from the trunk to Asterisk, where i try to pick up the phone.

I hope that you can give me any advice on what the problem is, as I really want this beautifull piece of software to work.
Link for log:
https://dl.dropboxusercontent.com/u/697024/itavis/full

-Sofus

Hi,

I am new to FreePBX and try to learn more about this great software.
I just downloaded and installed FreePBX Distro (4.211.64-9).
In the documentation (first steps after installation) it says that I have to configure Sendmail/Postfix.
As I can see Postfix is the default installation this distro.
What has to be configured in Postfix - I just want to be mailed by FreePBX about new updates and error alerts from FreePBX ?
I have entered my email address in FreePBX:

1. System Admin/Notifications Settings
2. Module Administration/Upgrade Notifications

Appreciate all answers ....

Let's see if I can get an answer without an argument....

I'm changing out my network to support VLAN per recommendation by SkyKing. Everything works, I can get all the phones on the VLAN but when I go to change the IP address of the FreePBX using the GUI, It says it changes but it really doesn't. The original address was 192.168.13.13. I change that to 192.168.1.13 and hit save, I still have to use 192.168.13.13 to access it and it still shows 192.168.1.13 in the address field.

If I reboot the machine, there is an error while shutting down that says my MAC address doesn't match, but it's ignoring that. However, when it comes back up, it says it can't assign ANY address to my eth0 adapter. I have manually punch in the 192.168.13.13 through command line before I can access it again.

I was soooo close to having this finished. I had to change everything back to no VLAN so we could continue functioning.

What am I missing?

Thanks for being....kind....

Hi,

I believe I've met the requirements for install of Sysadmin module, but it hangs during the installation procedure.

Http log shows this:

[Sun Dec 08 00:12:18 2013] [error] [client xxx] PHP Fatal error: Incompatible file format: The encoded file has format major ID 4, whereas the Loader expects 5 in /var/www/html/admin/modules/sysadmin/install.php on line 0, referer: http://xxx/admin/config.php

PHP info:

[root@xxx ~]# php -v
PHP 5.4.22 (cli) (built: Nov 13 2013 09:41:55)
Copyright (c) 1997-2013 The PHP Group
Zend Engine v2.4.0, Copyright (c) 1998-2013 Zend Technologies
with the ionCube PHP Loader v4.4.4, Copyright (c) 2002-2013, by ionCube Ltd., and
with Zend Guard Loader v3.3, Copyright (c) 1998-2013, by Zend Technologies
[root@xxx ~]#

From Zend Guard Release notes:

Could you please advise?

Thanks
T

Hi,

I have recently been trying to connect a trunk to a local provider but he requires IP authentication and is not able to give me ANY help with configuring my PBX.
He just gave me the following table and no more.
If anyone can help me configure a trunk and explain the info he has sent that would be greatI have read posts about this but none had reference to so many IP parameters.

5060 Sip Signaling Port
10 SIP Channels and Bandwidth
IP Address (Signaling) –SONUS1 XXX.XXX.66.43 signaling
IP Address (Signaling) –SONUS2 XXX.XXX.67.43 signaling
IP Address (RTP) –SONUS1 XXX.XXX.66.33
IP Address (RTP) –SONUS2 XXX.XXX.67.33
RTP PORT 10000-20000
Voice codec ALL CODEC
Fax protocol-FAX RELAY T38 AND FALLBACK TO G711
DTMF RFC2833 OR INFOOR BOTH
ROUTED TO : XXX.XXX.XXX.XXX
0732-0XXXXX

I need some custom work for Free PBX.

I would like to have user login to queues, it should ask them for a user and password. if verified, it should play a message there name and status (if they are logged in or out)
to login press 1 to logout press 2

Pressing 1:
Add Record:
Database: Asterisk
Table: Queues_Details
Fields: ID=8101, Keyword=member, data=Local/[userinfo]@from-queue/n,0, flags=0
Now it needs to run the cmd apply config.

Pressing 2:
Delete this users record from this server.

The phone is in SIP mode.
The freePBX server recognizes the phone is "offline".
When the sip debug is enabled I get this:

--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.105:50638 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From:
To:
Call-ID:

Date: Tue, 10 Dec 2013 06:07:39 GMT
CSeq: 101 REGISTER
User-Agent: CSCO/5
Contact:
Content-Length: 0
Expires: 120

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.105:50639 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From:
To:
Call-ID:

Date: Tue, 10 Dec 2013 06:07:39 GMT
CSeq: 101 REGISTER
User-Agent: CSCO/5
Contact:
Content-Length: 0
Expires: 120

I get an error on the phone E630.
The phone allows me to make phone calls out but of course not in.

Hi,
I've just installed freepbx distro downloaded today, install was all successful, only problem is that java ssh is not working. tried uninstalling and reinstalling ssh from but still no joy, the page loads in firefox but the security warning window never shows.
I've got another box with asterisknow/freepbx and can ssh into it without problem.
being new to freepbx I'd be grateful if someone can help me this.
thanks in advance.

PBX 2.0.6.4 Asterisk 11.6.0 FreeBPX 2.11.0.11 Dahdi 2.7.0.1
About 72 hours old

I know I have forgotten something here, but I need someone to hit the CTL+ALT+DEL on my brain please --

I have a system that is supposed to record all inbound and outbound on five extensions and the recordings were in the var/spool/asterisk/monitor directory, now they seem to be in /var/spool/asterisk/monitor/YYYY/MM/DD/ --- An I only see files that begin as OUT- So I am not sure I am getting inbound calls.

On the old box PBX 2.7.5.6 Asterisk 1.8.5 FreeBPX 2.8.1.4 Dahdi 2.4.1.

All the recordings were in /var/spool/asterisk/monitor an started with OUT- or the YYYYMMDD- formats depending on rather they were inbound or outbound calls.

Access was always through the ARI client with the admin + password.

But now I can't even get logged on with the admin + password.

Even after passwd-ari and confirming the password in amportal.conf

I have looked at a number of threads most import these two --

http://pbxinaflash.com/community/index.php?threads/authentication-required-on-user-portal.13204/page-2#post-84075

http://pbxinaflash.com/community/index.php?threads/ari-requires-username-password-maint-user.13935/#post-89055

Any HELP ???

TIA --

Hello

I have installed asterisk 11.6, 2.8 dahdi, libpri 1.4.14 and freePBX 2.11 in debian 7.0 and after making several updates I get the following error:

retrieve_conf failed, config not applied
Ignore this
Reload failed because retrieve_conf encountered an error: 126
Added 2 weeks, 4 hours, 24 minutes ago
(freepbx.RCONFFAIL)

Please I need help I'm new.

Regards, several days ago and then install freepbx upgrade to a more recent version, after that I could not install some modules the mistake is as follows:

The following modules are disabled because they are broken:
callback, callforward, callrecording, callwaiting, dahdiconfig, daynight, dictate, fax, javassh, languages, manager, pbdirectory, phonebook, setcid

You should go to the module admin page to fix these.
Added 0 minutes ago

So I built a fresh install with FreePBX distro over the past weekend (12/7/13), and found that our new Digium A8B card required DAHDI 2.8. I realized it was in beta, but I was up against a deadline - so I downloaded the 5.211.64-1 ISO, installed it, and everything seems to be running fine.

Now I see that there's stable version 5.211.65. I've got SysAdmin Pro, and several other modules. Is there any difference other than the version number in the release name? I know it says .64 - which means CentOS version 6.4, but the ISO installed CentOS version 6.5 (final).

Is there any way to update the core system without having to format and start again? If there isn't, can I at least backup my settings?

Sorry, I'm pretty new to all this. Thanks for any assistance.

Hello sir, How to implement FreePbx with asterisk on Centos 6.X. I've need to complete step to deploy on local server. Thanks & Regards, Reetaes