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Not Found The requested URL was not found on this server. -FREEPBX

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@drvirus wrote:

Hey Folks .
have freepbx 12
asterisk 13
##########
enabled http and AMI
i can login using AMI over telnet .
but not over http

htttp:.x.x.x:8088/asterisk/rawman?action=Login&username=admin&secret=secret5
always give me :
Not Found

The requested URL was not found on this server.

Asterisk Server

#############################
vps1460851189*CLI> http show status 
HTTP Server Status:
Prefix: /asterisk
Server Enabled and Bound to 0.0.0.0:8088

Enabled URI's:
/asterisk/httpstatus => Asterisk HTTP General Status
/asterisk/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/asterisk/static/... => Asterisk HTTP Static Delivery
/asterisk/ws => Asterisk HTTP WebSocket

Enabled Redirects:
  None.
vps1460851189*CLI> 
####################################
[root@vps1460851189 ~]# cat /etc/asterisk/manager_additional.conf 
;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
[cxpanel]
secret = cxmanager*con
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate
[xp10]
secret = xp10
deny=0.0.0.0/0.0.0.0
permit=188.6.19.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate
webenabled = yes 
[root@vps1460851189 ~]# cat /etc/asterisk/http_additional.conf 
;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
[general]
enabled=yes
enablestatic=yes
bindaddr=0.0.0.0
bindport=8088
prefix=asterisk
[root@vps1460851189 ~]# 

any help ???

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Cisco Phones Not Working on Conference After Updating SIP Firmware on Freepbx:

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@ArshadHussain wrote:

Hi Everyone,

Good day!.

I would be really appreciate you as you already configured Cisco 79xx series phones with CONFERENCES, recently we have implemented asterisk and we have configured 7962,7931 Cisco Phone SIP firmware via TFTP everything are working fine except conferences.

I am using SIP firmware on the following cisco phone model.
All are connected and working fine just not working on conference.
6921, 7962, 7931, 3905.

My humble request you to all that could you please help me out on this.

Please waiting for your kind response.

Thank you in Advance.

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FreePBX Distro Conversion Tool - v13 Paid Modules

What is the reason behind error " Cannot Connect To Asterisk "

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@siddharth_khade wrote:

Hi ,

My server was running fine but now a twice time i had error "Cannot Connect To Asterisk"
after restart the " service asterisk " I wanted know root cause

Please help for find the root cause .

Many thanks

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2 Systems, IAX2 Trunk - Can only call 1 way

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@mvogel4949 wrote:

I have two systems connected by an IAX2 trunk. Dial patterns are perfect. I can call from B to A but calls from A to B get a busy message and the call does not reach B. Any ideas?

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Any ideas - Fresh install - CPU pegged

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@kristiandg wrote:

Good afternoon all. Have an odd one and not sure where to look. Fresh install, then a database restore of our template database. Yum Update shows no updates, and all modules are on latest (non-edge) versions. However, CPU has been creeping it’s way up to 100% since it was rebooted this afternoon, and an HTOP shows it’s a MYSQL Daemon that’s hogging the cycles. :frowning:

Thoughts?

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Limit the ring time when we are calling on any cell phone no or mobile no

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@rajesh123 wrote:

How can i limit ring time when i am calling on cell phone no and mobile no if user not answering with in 25 sec or 15 or(whatever i can put) in freepbx ring time from advanced settings.

i configured ring time from asterisk advanced settings it works but when i am calling on any extension and making that extension terminate call if found no answer with in 25 or 15 or(whatever i can put in freepbx ring time from advanced settings)

please help me out .

thanks in advanced.

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SSL Cert and Sangoma Phones

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@dicko wrote:

I read on my news feed that Comodo/Trustico revoked 23000 certificates yesterday , where you unlucky?

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Queue Dial

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@bajramia wrote:

Hi Team,
I have create 2 Queues i have assign dynamic Agents 8802,0
when i dial the queue 8894* wont work it gives me a reorder tone

I need to be able to login to the queue as Agents from different location / extension

Thank you

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AdHoc Device & User Mode Voicemail MWI Notifications

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@asicforensics wrote:

Community,
I wanted to post this to see if anyone had any ideas / see if it could be fixed. I am using
Current PBX Version: 14.0.2.11
Current System Version: 12.7.4-1802-3.sng7
Asterisk 15.1.1
Voicemail module: 14.0.1.19

Our system is configured with Device & User mode enabled and each device being set to AdHoc mode. The devices and extensions work perfectly except for one problem. The problem I am running into is my decies are not receiving MWI notifications when new voicemail is delivered. I believe this is because the devices are not set with the mailbox field (since multiple users use a single device). The phones appear to be registering for noticications from the device instead of from the user (extension) when it is logged in. The phones work properly when the mailbox field is filled in with a user@default option, but not when left blank.

I was wondering if there was a way to have the phone still register to the device and have the device retrieve the mwi status from the logged in user? In this way the phone would not have to re-register each time a user logs in or out; and it could provide the correct voicemail information on the phone. I don’t know enough to know which files need to be adjusted for the mwi settings, but would be grateful for any assistance. Currently we have to have the system send out email notices when a message is left because there is no notification on the phone.

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Homebrew freePBX cloud with ATA

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@aquitain3 wrote:

I have residential analog infrastructure and magic jack. Looking to upgrade to a more powerful system with minimal cost.

I am thinking about going down one of two routes:

1: Buy the rasberry pi, an ATA, and use freePBX with a cheap sip provider.

2: Skip the Pi and put the PBX in the cloud instead of on a local box.

I have never used an ATA, but I am assuming that it will be configurable so that I can use the cloud PBX? Also, I am concerned about reliability and stability issues. I have Verizon FIOS.

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Exp100 Expansion Module - How To Brighten Up Display

Turn off tone when call is put in parking lot

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@Bradbpw wrote:

When a call is placed in the parking lot all the Sangoma phones in our office will make a short tone. I have been trying to find a way to disable this but can’t find the setting. According to this feature request, it should be under REST APPS Settings, but I can’t find those settings. Where would I go to disable this?

I’m using FreePBX 14.0.1.36

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Fwconsole restart fails into restart and asterisk remains shutdown

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@xptpa2020 wrote:

FreePBX v13 updated to most recent release. Updated with latest modules in edge.

We issue an fwconsole restart weekly (srtp issue requires periodic restart). Now fwconsole restart fails after starting command and leaves asterisk in not running state. Output reads as such:

[root@FreePBX ~]# fwconsole restart
Running FreePBX shutdown…

Stopping UCP Node Server
Stopped UCP Node Server
Stopping Chat Server
Stopped Chat Server
Stopping Zulu Server
Stopped Zulu Server
Shutting down Asterisk Gracefully. Will forcefully kill after 30 seconds.
Press C to Cancel
Press N to shut down NOW
[============================] 1 sec

[Exception]
Hook file ‘/var/spool/asterisk/incron/firewall.stopfirewall’ was not picked up by Incron after 5 seconds. Is it not running?

restart [-i|–immediate] [args1] … [argsN]

Must then issue an fwconsole start, while it does start, there is a noted exception:

[root@FreePBX ~]# fwconsole start
Running FreePBX startup…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
25213 [============================]
Finished setting permissions
Running Asterisk pre from Firewall module
Hook file ‘/var/spool/asterisk/incron/firewall.firewall’ was not picked up by Incron after 5 seconds. Is it not running?
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Restarting fail2ban
fail2ban Restarted
Updating License Information for 17497836
Checking Vpn server
Starting Asterisk…
[============================] 5 secs
Asterisk Started
Running Asterisk post from Endpoint module
Running Asterisk post from Pagingpro module
Running Asterisk post from Ucpnode module
Starting UCP Node Server…
[>---------------------------] 2 secs
Started UCP Node Server. PID is 11436
Running Asterisk post from Xmpp module
Starting Chat Server…
[>---------------------------] 1 sec
Started Chat Server. PID is 11559
Running Asterisk post from Zulu module
Starting Zulu Server…
[>---------------------------] 1 sec
Started Zulu Server. PID is 11628

Concerned that restart fails. Concerned that a start yields a “firewall.stopfirewall was not picked up by Incron…”

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Dialogic Diva Forum?

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@Dave_D wrote:

Hello,
I got sent here from the Dialogic Developer’s Forum. I don’t see any existing Dialogic related posts. Wondering if I’m in the right place… please advise.
Thanks,
Dave

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Transfer call to busy extension

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@wassy83 wrote:

Hi to all I cannot figure out this:
external call is ringinging to call group 1010
extension 101 pick up the external call
extension 101 wants transfer the external call to extension 102
extension 102 is busy
extension 101 make a blind transfer to 102
the caller is listening the busy tone and asterisk will terminate the call

what I want is to put the caller on hold forever until extension 102 will be available again.

call waiting is disabled cause my colleagues don’t want to listen the free tone if an extensions is busy.

any suggestion?

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Is it possible to make FreePBX PCI compliant

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@kristech wrote:

We are looking to implement FreePBX in our call centre but before we make this decision I would like to know if it is possible to make FreePBX PCI compliant?
Any info on this topic will be greatly appreciated.
Thanks.

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Call Accounting / CDR reporting suggestions

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@munozj wrote:

I’ve used Call Accounting Mate for years but they haven’t developed on it since 2006. What is everyone’s faviorite Call Accounting software now-a-days?

Trying to get more statistical information about calls and queues instead of the raw CDRs.

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Sip Registration fails often

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@MacsOffice wrote:

I’m new and have freepbx working using e4sip. I have had 3 instances in the last 2 weeks when the sip registration fails at the provider and of course the system doesn’t work. I have reported it to them each time and by the time they reply it had started working. and they say nothing is wrong.
Have I picked a bad provider or is there a way to find out what the sip registration failing is caused by? I’m connected through Comcast Business and I have asked that they not block any ports and they said that has been done. The fact it works at all I suppose means they are correct but why is this failing so often the same way.

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Mitel M-5330e - Endpoint Manager

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@DanielHarker wrote:

Recently had to get a Mitel M-5330e working with the Commercial Endpoint Manager. The M-5330 is listed but not the M-5330e, which is identical but supports gigabit Ethernet. Annoyingly, the Mitel file format includes the model number on the first line and the phone rejects the file if it does not have the e in it. So I had to find a way to add the M-5330e.

Here’s what I did:

  1. Use MySQL Workbench to connect to the MySQL instance on FreePBX using the root credentials

  2. Open endpoint_models and sent the following command:
    INSERT INTO asterisk.endpoint_models
    (brand,type,model,accounts,prgkey,softkeys,topsoftkey,extrakey,linekeys,horsoftkeys,speeddial,global,firmware,exp)
    VALUES
    (‘mitel’, ‘phone’, ‘M-5330e’, NULL, NULL, ‘24’, NULL, NULL, NULL, NULL, NULL, NULL, ‘’, ‘0’);

  3. Open endpoint_basefiles and use the following command:
    UPDATE asterisk.endpoint_basefiles set model = concat(model, ‘, M-5330e’) WHERE brand=“mitel” and model LIKE ‘%M-5330%’;

And now the 5330e will appear as an option along side the 5330.

The Mitel phones seem a bit awkward in terms of getting them to call the TFTP server in the first place, thus far we’ve only been able to get it to work by manually setting the TFTP server in the phone interface.

Just thought it might be useful information to share.

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