Configure: error: xml2-config not found

configure: error: xml2-config not found. Please check your libxml2 installation.

Reference : http://asherbond.com/blog/2009/05/05/configure-error-xml2-config-not-found-please-check-your-libxml2-installation/

How to give a standard Ubuntu User full root permissions.

Reference : http://bobbyallen.wordpress.com/2007/01/15/how-to-give-a-standard-ubuntu-user-full-root-permissions/

How to Setup TFTP on Ubuntu 11.10

Reference : http://icesquare.com/wordpress/how-to-setup-tftp-on-ubuntu/

Follow me settings does not work with Mobile numbers

OS : Free PBX 2.10

Issue : I have added follow me setting to one Extension and it included internal extension and Mobile number. These settings only work for Internal extension.

Solution : I have added Mobile number followed by # then it worked fine. Both Internal Extension and mobile number rang for follow me.

488 Not Acceptable with Grandstream Phone and Free PBX 2.10

If you got this kind of error then you should check the codec that used by the phone and what are the codec that were enabled from the server.

First you can view through the phone display and make sure the codec that phone used.

In Free PBX 2.10 version you can move to Settings ----> Asterix SIP settings -----> Audio Codecs

and make sure phones supported codecs are enabled.

This is almost issue with Codec.  

SIP server Outside Call access

OS : Free PBX 2.10

Requirement : I have FreePBX running VOIP server and one of ISP has provided SIP trunk. VOIP server has run without issue and incoming and outgoing calls are working fine. SIP trunk doesn't have any static IP. But we do have static IP with another ISP. And that link also plugged into the VOIP server.

I need to configure SIP phone to make calls through my VOIP server while they are out of the network (Out side.)

There are several steps that you must follow to do such configuration.

1. All Signaling and Media ports should forwarded to Asterisk

   UDP and TCP 5060 port

    2. All Signaling and Media ports should forwarded to Asterisk.

    3. The Extensions/Devices is setup to be NATed.

This confugration should done only if both SIP server and client behind the NAT.

In order to accomplish the above we need to apply some configuration information into FreePBX, some Asterisk configuration files and on your firewall/router.

Internal/External Network Information

You must edit or create the file sip.conf typically found in your /etc/asterisk directory and make sure it is owned by asterisk. We will assume that you have an internal network of 192.168.1.0/255.255.255.0 and that you have a static IP address of 24.72.182.16. If you have a dynamic IP, see the notes that follow. In this situation, you need to create or edit the following entries in your sip.conf file:

Nat=yes

externip=24.72.182.16
localnet=192.168.1.0/255.255.255.0

This tells Asterisk what IP address range is internal vs. external so that it can rewrite the SIP headers appropriately. If you have a dynamic address instead of a static address then you need to modify the above. You will need to have a domain name for the host, let’s assume you are using dyndns.com’s free service and have chosen the name mydomain.dyndns.org. Then yoursip.conf file would look like the following:

externhost=mydomain.dyndns.org
externrefresh=120
localnet=192.168.1.0/255.255.255.0

Nat=Yes

Where externrefresh tells Asterisk to recheck the IP address every 120 seconds in this case. You should adjust this higher or lower based on the frequency that this changes.

Firewall/Router Configuration

The default installation of FreePBX is configured to use UDP port 5060 as the SIP signaling port and UDP ports 10001-20000 as the RTP Media ports. All these ports must be forwarded to your FreePBX System. How to do this varies widely depending on the firewall or equipment that you are using. It is commonly referred to as Port Forwarding or maybe Destination NAT (DNAT). However it is referred, if we assume in this example that your FreePBX system has an internal IP address of 192.168.1.100 then you will want:

• UDP/5060 -> Forward to 192.168.1.100
• UDP/10001-20000 -> Forward to 192.168.1.100

Extension Information

We will assume you are using FreePBX in Extension mode but if you are using Devices/Users the same applies on the Devices page. You need to configure the extension with NAT enabled so that Asterisk knows this device is NATed and can apply the SIP rewriting rules that you previously configured in the sip.conf file. Navigate to the desired extension and scroll down to the Device Options Section.

The configuration option nat must be set to yes, and you may want to set qualify to yes as well although not necessary.

With these steps, when properly configured, your external device should be able to communicate with your FreePBX server unless you have issues on the remote end where the device is located because of badly behaved Firewalls. The remote device should be configured to use your external IP address or domain name as configured above in the sip.conf file.