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multiple premium numbers

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Dear all,

hope are all ok

I would like some assistance, am having a current freepbx with one e1 and using digium single span card and using pstn connection from a Telecom company. I am running premium services and would like to configure 2 numbers/lines on one e1, how can I do this such people can call number 1 and 2 at the same time but with each number having different service.

I am also using follow me feature to divert calls to about 10 different mobile numbers is it possible for more then one call to come through to different numbers is one of the numbers is already engaged?

I will be greatful to any assistance, am a newbie.

thanks

Yusuf

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FreePBX and Cologne Chip Designs GmbH ISDN network controller [HFC-PCI

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Dear PBX'rs!

I have recently replaced a [working] debian/lenny server with an ancient Asterisk installation with a debian/wheezy system, latest asterisk (1:1.8.13.1) and FreePBX 2.11.0.11 connected to a UK BT ISDN2e line with the 'Cologne' ISDN card.

I have started completely from scratch and not attempted to reuse any of my old config files.

After a few hours of hacking I now have the card detected in the DAHDI module:

Span: Cologne Chips - HFC-S PCI A ISDN card 0 [TE] [1]
Alarms: OK
Framing/Coding: CCS/AMI
Channels:3/3
D-Channel:3
Signaling: pri_net

I also have a [single] working SIP extension (e.g. '*43' works as an echo test)

but I'm now a little stuck as to how to proceed with trunks and in/out routes. I've read various forum posts from earlier this year implying that maybe this card isn't yet fully supported in which case I'm on to a loser.

A little gentle guidance in the right direction would be very greatfully received.

Thank you,

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/var/run/asterisk/asterisk.ctl apparently does not exist.

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Hi All,

I am attempting to install FreePBX (version 2.11.0) on top of Asterisk (1.8.20). The OS is CentOS 6.4 Final. These are all running on a VPS.

My problem is that FreePBX says that Asterisk is not running but I can log into the CLI using asterix -r. This is the output from a few commands:

1.
[root@darkstar ~]# service asterisk stop
Stopping asterisk: Asterisk ended with exit status 0
Asterisk shutdown normally.
[ OK ]

2.
[root@darkstar ~]# service asterisk start
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Starting asterisk: [ OK ]

3.
[root@darkstar ~]# asterisk -cvvv
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect.

4.
[root@darkstar ~]# asterisk -rvvv
Asterisk 1.8.20.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.20.0 currently running on darkstar (pid = 1795)
Verbosity was 0 and is now 3
darkstar*CLI>

As you can see, asterisk startup reports that /var/run/asterisk/asterisk.ctl may be missing and then asterisk -cvvv reports it is already running using asterisk.ctl and the CLI is accessible.

I have searched everywhere (well probably missed a few places Smile but can't seem to find a solution. I have checked the file exists. I have checked that the admin password is the same in both /etc/asterisk/manager.conf and /etc.amportal.conf.

Any clues as to what else it could be would seriously be appreciated. Thanks for reading about my problem.

Regards

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freepbx 2.11.0 giving me php errors when installing and when I try to go to admin page.

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HI all.. I'm trying to install freepbx 2.11.0 with asterisk 2.8 on a linux server running ubuntu 13.10. the thing is, is that I did several reinstalls, both with asterisk and with freepbx, trying different versions of freepbx but it is not working for me. I get the following error wen trying to visit my web interface of freepbx:
Warning: include_once(/etc/asterisk/freepbx.conf): failed to open stream: No such file or directory in /var/www/html/admin/config.php on line 106

Warning: include_once(): Failed opening '/etc/asterisk/freepbx.conf' for inclusion (include_path='.:/usr/share/php:/usr/share/pear') in /var/www/html/admin/config.php on line 106

Fatal error: Call to undefined function module_run_notification_checks() in /var/www/html/admin/config.php on line 139. Does anyone know what the probly is? If someone could help out with this, I would be very grateful. Thank you all.

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Provisioner.net is down, need tgz files

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I'm in the process of setting up a new PBX for our company. We have ordered several phones for vendors to see which we like. I'm unable to load the endpoints config files from the configuration manager. I've also tried to get the tgz files from provisioner.net and load them directly but the wiki is down with a database error.

I sent an email to Andrew Nagy several days ago but have not heard a response back. I'm under a time crunch and need to get started on this project. Can any one offer some help in trying to get these files loaded - right now we'ld like to test polycom and yealink.

Any and all help is appreciated. Tks.

(if this is posted to wrong forum, I apologize)

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Calls being cut off after 6 seconds

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Hi,

I have a phone that is connected over the internet to FreePBX. That phone -and that phone only- cuts off calls after 6 seconds. It's quite weird that we have another phone is connected over the internet fine.

No idea what is going on- any ideas?

Thanks,
Sam

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FreePBX using Optimum Cannot get SIP working

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I need help configuring Optimum SIP and Asterisk.

I have Optimum SIP to a modem and have the FreePBX box connect to it but i am not able to receive or make calls.

the register string is: 9143584818:9143584818@10.40.7.202/9143584818

PEER details
host=10.40.7.202
type=friend

thanks for the help
michael

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Intel 82574L onboard ethernet

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May be somebody has come accross this and got a solution, as i have sent hours on google and this forum trying to find a fix.

Got a supermicro X7SPA-HF-D525-0 MB with dual nic onboard i have tried to install with the latest image and get a kickstarter error, so i tried to configure the nics, select the driver e200e and then get an error saying it cant find hardware, downloaded drivers from intel to a USB drive and tried installing from there same error.

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Menus & buttons not showing correctly after upgrade to 2.10.1.10

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Hello all,

I had a running system which was running fine until today when I upgraded everything using the "Module Admin" on freepbx. The version which I was running was 2.10.1.2. After performing an upgrade to 2.10.1.10, I noticed that the menus are not showing correctly, buttons are not shown at all and basically the user interface is out of whack. I tried different browsers, chrome, safari, firefox, IE and they all came out to be the same. I can not find a solution for this so I'd appreciate your help regarding this.

Thanks

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International Calls?

sysadmin installation fails / hangs

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Hello

I am setting up a debian box with the freepbx tarball (2.11). Everything looks quite ok, I can make calls (asterisk 11.6).

However, installation of module system admin fails. I have satisfied dependencies on Zend Guard Loader & incrond, but when I try to install, I get the status windows "Please wait while module actions are performed" - but nothing happens.

I've tried to execute the install script directly (calling in the url modules/sysadmin/install.php) and got a message back saying that the script cant be access directly. Which leads me to believe Zend Guard is ok.

1- What could be wrong ?
2- I'd like to re-download the module (currently, module admin says it is local and can be installed.). How can I revert to "download" status ?

My first question is obviously what I really need !

Thanks for any support you can bring

J

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Issue with configuring phones

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Let me start this by saying I am new to FreePBX and VoIP in general. My company wants me to get FreePBX set up and running as a possible alternative to what we are currently using. I was able to get FreePBX installed and running, and was able to make inbound and outbound calls on my softphone (x-lite.) The issue I'm running into is I cannot get FreePBX ( with end point manager ) to configure my polycom ip560 phones. When I reset the phone I get an error message that says " cannot contact boot server, using previous configuration " and the phone goes back to how it was configured before I tried to mess with it.

I'm sure at this point that it's just user error, but i'm not sure where I have made the error. I'm hoping someone can at least point me in the right direction here. I'll start with how I have EPM set up.

Global Settings:
Internal IP Address: (IP address of freePBX system)
External IP Address:
Web Server Port: 80
HTTP Provision Port: 84
RESTfull Apps Port: 88
SIP Port: 5060
Phone Admin Password: 000000
Phone User Password: aa0000

Template for polycom:

Template Name:? tpi
Destination Address:? Internal
Missed Calls:? Enable
Persistent Volume:? Enable
Call Waiting Signal:? Beep
Time Server:? time.schmoozecom.net
Daylight Savings:? Enable
Time Zone:? -6
Background Image:? logo.jpg
Provision Server Address:? Server Address: (ip address of FreePBX)
Provision Server Protocol:? TFTP
Line Label:? Name Extension
Firmware Version? Recommended

On my actual Ip560 Phone, and this is most likely where I am doing something wrong.

DHCP Client: Enabled
Boot Server: Custom+Opt.66
BootSrv Opt: 160
Bootsrv Type: String
Vlan discovery: Fixed
Vlan ID opt: 129
IP Gateway: 192.168.1.143

Server Type: Trivial FTP
Server address: IP address of FreePBX
Sever User: 000000
Server Password aa0000
File Tx Tries: 3
Retry Wait: 1
Tag SN to UA Disabled
SNTP Address blank
GMT Offset -6
DNS Server 192.168.1.62
DNS Alt. Server: 000.000.000.000
DNS Domain : Blank

I really hope someone can help me out here. Thanks in advance

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new installation

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Here are my computer specs:
CPU: intel i5
RAM: 8GB
MB: Gigabyte Z77MX-D3H
HD: WD20EARX-0

I have downloaded the latest freepbx 64-bit ISO and begin the installation...sadly I only get to the Probing EDD ok screen but that's as far as I get....I tried searching the forum for this error but no luck....perhaps I am the ONLY one to have this error. Any help would be appreciated

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failed for Freepbx distro

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Hi ..
I am doing installation of freepbx distro from iso version "FreePBX-4.211.64-8-i386-Full-1384947526". The installation get complete without any error but when I tried to open the Webpage to configure the Extension it gives me APACHE page.
Could you please help me.

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Google Voice (Motif) - Wierd problem


Asterisk issues / assume NAT problem, need help

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The Asterisk/FreePBX server is in our data center behind our firewall. It's on a 10.10.40.0/24 network. Our telephones are at a remote site that has an IPSEC tunnel to the data center. These devices are on a 10.1.40.0/24 network.

Calls between the phones at the remote sites work. Calls from our remote site to an extension on our Asterisk server (Directory for example) do not work; calls via our SIP trunk provider configured on the Asterisk server also do not work - there is no audio.

I'm sure this is a beginner NAT issue; I'm looking for guidance on how to resolve.

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Help Dialing outside line

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Hi,

I wonder if someone could give me a bit of help with the dialling plans. Basically I can call internally, but when I try and dial out it just wont. the sip account is registered, I just cant seem to figure out what the correct dialling plan should be.

Thank You in advance

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Skype Connect

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I have a working Piaf green 64bit running, and I am trying to setup my skype connect account as a sip trunk. It is hard to find information or guides on how to do so. I was able to get outgoing calls to work, but I can not get incoming calls to work. I am new to asterisk and freepbx, so please be gentle!

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Cannot access user control pannel. Wrong path.

No In-call after upgrading freepbx 2.11

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Hello,

I just upgrade freepbx to 2.11 version, and I loose all incoming call.
I only could make outside call.
All my trunks looks ok.
I only get a problem in this module
System Admin 2.11.0.7 Schmoozecom.com Disabled; Pending upgrade to 2.11.0.35

Is it important to get this module OK?

I use Sip from OVH.

Yann

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