@chenboly wrote:

Hello, I am new in this forum as well as new to use Elastix, I recently restore from my backup file, however during the restoration, the error message of:

"INFO: restoring component endpoint/ep_db...
FATAL: endpoint/ep_db: failed to read list of MySQL schemata: SQLSTATE[28000] [1045] Access denied for user 'root'@'localhost' (using password: YES)" is appeared and the restoration is not successful.

After the unsuccessful restoration, my elastix got error when I access to PBX Configuration page.

Before I posted to ask for help, I try to resolve follow by the some solution had posted in this forum too, but still no luck.
The issue that I am still cannot figure out is I could not see any database and the root access also denied to show any MySQL database.

Please kindly help me and here is the error:

FATAL ERROR
DB Error: insufficient permissions
Trace Back
/var/www/html/admin/libraries/db_connect.php:75 die_freepbx()
[0]: DB Error: insufficient permissions
/var/www/html/admin/bootstrap.php:85 require_once()

and much more error message..

here is my detail of Elastix:

Kernel
Linux(x86_64)-3.10.0-229.14.1.el7.x86_64

Elastix
elastix-4.0.0-1
elastix-a2billing-2.2.0-0
elastix-a2billing-callback_daemon-2.2.0-0
elastix-addons-4.0.0-5
elastix-agenda-4.0.0-3
elastix-asterisk-sounds-1.2.3-1
elastix-email_admin-4.0.0-6
elastix-endpointconfig2-4.0.0-3
elastix-extras-4.0.0-3
elastix-fax-4.0.0-2
elastix-firstboot-4.0.0-3
elastix-framework-4.0.0-18
elastix-im-4.0.0-2
elastix-my_extension-4.0.0-2
elastix-pbx-4.0.0-8
elastix-portknock-0.0.1-0
elastix-reports-4.0.0-6
elastix-security-4.0.0-4
elastix-system-4.0.0-11

RoundCubeMail
RoundCubeMail-0.3.1-12

Mail
postfix-2.10.1-6.el7
cyrus-imapd-2.4.17-13.el7

IM
openfire-3.7.1-2

FreePBX
freePBX-2.11.0-26

Asterisk
asterisk-11.25.0-0.el7.centos
asterisk-perl-1.03-0
asterisk-addons-11.25.0-0.el7.centos

FAX
hylafax-4.3.11-1rh7
iaxmodem-1.3.0-0.el7.centos

DRIVERS
dahdi-2.10.2-0.el7.centos
rhino-0.99.7-0
wanpipe-util-7.0.14.3-0

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@dmanolis79 wrote:

Just installed a new FreePBX 14 SNG7, And I noticed that the server makes about 8 DNS requests every 15 seconds. I have other servers installed and they dont make any requests. Is there a way to find out why its making those requests frequently?

thanks in advance.

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@mixhali wrote:

Hi,

I'm not really that new to NAT in fact know it pretty well however I'm newbie with how Asterisk and the SIP protocol handles NAT

Ill explain my architecture a little first
I have a AsteriskNow box running version 13.0.192.16.
It sits behind a pfsense firewall that has NAT disabled. It simply forwards packets that meet valid rules.
On the pfsense box i have set up forwarding rules for SIP 5060 TCP/UDP as well as TCP/UDP ports 10000-10020
in front of my pfsense box is a cisco 1941 router. This is my edge device that has a static IP and handles my Nat requirements. I have set up nat rules on this box to forward sip/rtp ports to my asterisk box.

Problem, Typical issue, no audio for external x-lite clients.

X-lite clients register and can initiate a call. just no audio both ways. These are X-lite clients on iphones and i have tried using 4g data as well as wifi. Note they work fine when using wifi internally.

I have researched this to death and perhaps its just not ideal to do this but i just cant believe it. This has to be a very common requirement. Every time i find something on the net it either points me to adjust settings that simply don't exist on my version or don't help.

The one thing im sure of is that asterisk is not providing the correct details for the rtp connections. I can see the private IP address a lot in the x-lite logs which i suspect have some setting configured incorrectly that tells asterisk to send the correct IP info in the SIP messages.

This is a dump of show sip
MBICPBX01*CLI> sip show settings

Global Settings:

UDP Bindaddress: 0.0.0.0:5160
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-13.0.192.16(13.12.1)
SDP Session Name: Asterisk PBX 13.12.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: XXX.XXX.45.16:0
Externrefresh: 10
Localnet: 192.168.50.0/255.255.255.0
192.168.70.0/255.255.255.0
192.168.20.0/255.255.255.0

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en_AU
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
I've seen references that i need to enable Nat for the extension however i just dont see that setting in this version of AsteriskNow. I have seen screenshots of earlier versions where it is a clear yes/no option. I can only assume that it is redundant now for some reason.

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@alejoprivas wrote:

Greetings,
i have been trying to create an web app that connects with an webrtc client (jssip, sipml5 or sip.js) to my freepbx 14, all of them give the same result to Mozilla/5.0, even back tracked to chrome 49 and have the same issues.

Audio= works perfect both ways.
Video= softphone or hardphone receives video but browser wont show video.
dtmf= works both ways.

i tested jssip, sipml5, sip.js clients demos and all gave the same problem.

i tried this kurento asterisk tutorial from webrtc ventures (cant post links), but it crashes whenever i pick up the phone with this TypeError: this.mediaHandler.hasDescription is not a function.

every information about webrtc clients is atleast 5 months old so im not even sure is it possible to achieve.

im not forced to use freepbx 14, i could revert to asterisk 11, but still dont know if that would change anything.

this is my webrtc extension:

[2000] ;kurento-appserver
host=dynamic
secret=asterikpwd
context=from-internal
transport=ws,wss,udp
type=friend
encryption=no
avpf=yes
icesupport=yes
directmedia=no
disallow=all
allow=ullaw,opus
allow=vp8
videosupport=yes

im using chan_sip cause pjsip throws forbidden 403

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@Harpocrates wrote:

Will Icinga2 monitor a FreePBX13 server without issue? If so can the icinga2 agent be installed without issues w/ the epel-repository, and is it the preferred method?

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@bruceisg wrote:

Hi guys,

I am having issue with unlimitel trunk connections. the out bound call is success. but failed with inbound call. After the trunk debug, I could see the error code as below,

CAUSE : No authority found
CAUSE CODE : 50

my configuration for inbound:

disallow=all
allow=ulaw
canreinvite=no
context=from-pstn
type=friend

Register String: MYDID:MYPASS@iax06.unlimitel.ca

anybody could help or give some directions, much appreciated.

Thanks for any help.

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@Bradbpw wrote:

I am trying to install the default certificate generated at install time in the HTTPS Setup. In HTTPS Setup I select the "default" certificate and click install. The system hangs and I lose access to my GUI. The only way for me to get access back to the GUI is to SSH in and edit the file /etc/httpd/conf.d.ssl.conf. I have to edit lines 22, 39, & 54. On each of those lines it has:

ServerName Asterisk Private CA:xxxx

I have to change it to:

ServerName Asterisk:xxxx

Then restart httpd. That's the only way I can restore access to my GUI.

What am I doing wrong here?

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@julien wrote:

Hello,

in my company i have one queue with 4 agents (A) with a waiting of 13 minutes for the caller.
And i have another agent (B) who are not in the queue.

Actually i want that if after 30 secondes in the queue, nobody respond to the caller, then the caller will be transfered at the agent B... BUT if the agent B and Agents A not respond then the caller just wait 13 minutes in the queues (with A agents).

Thank you !

ps: sorry for my english !

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@mitterhuemer wrote:

Hello Community,

we have some problems with a sip trunk.

After a few days the sip trunk goes into state

"Rejected"

and does not try to re-register anymore.

I already tried to set the option "register_retry_403=yes", but this does not work in that case.

At the moment the only way is a

"fwconsole reload"

command to force a "re-register".

Registration Attempts is already set to "0" and the timeout is set to 20.

So in the case of a "Rejected" there has to be done something other to force a re-register.

Can someone help me please? At the moment i wrote a script, which runs as a cronjob, that checks the registry state and forces a reload when the state "Rejected" occours, but this is not really a solution.

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@shanegrogan wrote:

I just did a fresh install on a machine with 2 NICS. The GUI only shows ETH0 however. Console shows ETH1.

I would like to configure one NIC with an external IP and the other NIC with an internal IP.

Previous versions of FreePBX would show both NICS under system admin, network settings.

Thanks in advance.

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@karthick wrote:

To add the Paging or Intercom extension, I just insert data into following tables

Paging_config and Paging_groups

To add Paging extension, I just enter into the MySQL and give the insert query on the same PC, which has the FreePBX and Asterisk.

Is it possible to make MySQL Connection through Java application, on the SAME PC ? I need to insert data via my application.

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@mixhali wrote:

Is anyone using the above client with Asterisk? I just downloaded it and it is asking me to login. Not sure what i am logging into. I've tried putting an account that's valid on my box but im not sure that that's what i am logging into.

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@fetoa wrote:

Hi again,

We have realized that on internal calls, zulu is "rebooted" or "reconnected". This casuses that any text written on the chat is lost. But this doesn't happen on outbound calls. Is this a normal behavior?

Regards

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@fetoa wrote:

Hi again,

Whe can a found the websocket log of freepbx??? I need some information of the connections...

Regards.

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@traymaster wrote:

I am considering retiring my old analog system and replacing it with a Sangoma system. Sangoma offers a FreePBX unit and also the PBXact phone systems (already installed with additional modules?). I am looking for information on the way to proceed with this. And switching from analog to SIP now we have finally got fast broadband?

Any thoughts and advice welcomed?

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@Ajred wrote:

Hi I have Yealink t42s phones and need to put a button on the phone to toggle call recording on/off. I know I call *1 during call but users are asking for a button.

Is this possible?

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@yyy wrote:

Hi!

I've recently faced with strange issue. FreePBX distro 10.13.66-16 with mostly default contexts.
I have PJSIP-Trunk and 188 PJSIP-Extensions and all worked just fine until I've added another one PJSIP-Extension.
Incoming calls to Trunk began to be sent back to the caller's number, then came back, etc until the answer came busy.

All logs doubled many times
Part of the log of "one" call looked like this.

PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5
PJSIP/******-000000b6 is making progress passing it to PJSIP/****-000000b5

Then I deleted some extensions and everything was fine again. I started to add extensions one-by-one and when I reached the number of 190 PJSIP contacts at all(extensions+1 Trunk) - the same issue appeard.

Finally I've noticed that my Trunk had context "from-internal", and changed it to "from-trunk", and all calls became fine with any number of PJSIP contacts.

But why this issue appeared only when I reached 190 number of contacts? Is it bug?

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@fetoa wrote:

Dear all!

I wan to know if it's possible to create one system recording with many sound files of different languages. I've tried to create one system rcording, and tried to drag and drop different files in different languages but it's not possible. I have to create one system recording for each language.

Is there any posibility to create just one system recording and many recordings?

Regards.

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@VideoDudeMike wrote:

Sangoma Introduces Comprehensive Training with Sangoma University

Sangoma University offers a comprehensive selection of both Online and Classroom technical training courses for our partners and customers. Empowering telecom administrators, service integrators, and resellers with the necessary skills and experience to deploy and maintain Sangoma solutions. Follow several distinct programs to achieve recognition of your knowledge with Sangoma Certifications. Read the full blog post at https://www.freepbx.org/sangoma-introduces-comprehensive-training-with-sangoma-university/

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@fetoa wrote:

Hi all!

Does anyone tried to export/import extensions and queues from freepbx 13 to 14? Does it work properly?

Regards

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