Revision of Asterisk A Closer Look from Fri, 15/12/2006 - 6:51pm

We will start by downloading and compiling asterisk:-

---

Before we start we need to install some requirements I always install the headers and the development packages for mysql , postgresql and openssl in case I wanted to use any of this features with asterisk later on also asterisk require the kernel source, kernel headers, bison and libnewt then we will start by extracting and compiling: tar zxvf libpri-X.X.XX.tar.gz cd libpri-X.X.XX make su - make install

---

tar zxvf zaptel-X.X.XX cd zaptel-X.X.XX make su - make install

---

tar zxvf asterisk-x.x.xx.tar.gz cd asterisk-x.x.xx make su - make install make samples

---

tar zxvf asterisk-addon-X.X.XX.tar.gz cd asterisk-addon-X.X.XX make su - make install

---

tar zxvf asterisk-sound-X.X.XX.tar.gz cd asterisk-sound-X.X.XX su - make install and You are done with asterisk compiling..

---

now let us talk about some basics

1. asterisk configurations exists at /etc/asterisk/
2. asterisk sounds exists at /var/lib/asterisk/sounds
3. asterisk music on hold files at /var/lib/asterisk/mohmp3/
4. asterisk voicemail files and others at /var/spool/asterisk/
5. asterisk applications, codecs and channels at /usr/lib/asterisk/modules/
   1. sip.conf it contain the sip server options , sip registrar and the sip users
   2. iax.conf it contain the iax server options , iax registrar and the iax users
   3. extensions.conf it contain the dialplain which we call asterisk heart
   4. modules.conf it contain which module not to be loaded or loaded
   5. voicmail.conf it contain the voicemail options
   6. logger.conf it contain the logger options
   okay now let start with a very basic scenario .. you want to have 2 sip clients can reach each others so we need to have 2 sip accounts and a very basic dialplan. so we can move the sip.conf that already created from make samples and we keep it as a reference and we start or simple one from scratch

   
   [general]              ; here we will define the general options 
   
   bindaddr = 0.0.0.0     ; so the SIP server will be listing on all the interfaces
   bindport = 5060        ; default SIP port
   
   

   and now we will need to create 2 SIP account for our softphones username=1001 ; the username for authentication secret=123 ; the password type=friend ; what kind of user is he ? peer,user,friend context=test ; context is one of the key words it's where the incoming and outgoing calls from this user should go host=dynamic ; if the user don't have static IP address or he moving alot with his softphone nat=yes ; if user behind the NAT or no canreinvite=no ; this is important it define if the media path for the call should be directed between the end points or it should go via the asterisk server and almost the same configurations will be for the 2nd user username=1002 secret=123 type=friend context=test host=dynamic nat=yes canreinvite=no now the heart of this call the dialplan in extensions.conf

   
   [general]
   
   [test]                          ; the test context
   
   exten => 1001,1,Dial(SIP/1001) 	; when someone dial 1001 should dial 1001 using SIP protocol
   exten => 1002,2,Dial(SIP/1002) 	; when someone dial 1002 should dial 1002 using SIP protocol 
   

   now the easy part the softphone ... there is lots of softphones for both GNU/Linux and windows you can use xlite and adjust the username / authentication for each users the domain / sip server is the asterisk IP server and the password is the user password and we are done ;-) later one i hope this howto become more advanced but this is just a start