For Point 6, I found the problem is in my calling rules, I removed the 2 zeros, and it is working, but now I am getting in /var/log/asterisk/messages:
NOTICE[7230] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 127.0.0.1
Could anyone please help me in my previous problems, and this problem?

Thank you for your fast response.
I am trying to configure an Asterisk PBX at which I can connect from any place and make calls using my 2 trunks.

  Can you give more details on issue 2, and 4?
  For issue 1 I can resolve the domain name correctly from the outside network, and the domain name is the same like my server name.
   For issue 5, I am confused that the server is listening to the 5060 tcp port and not the UDP one, can you tell me why?

Can you please explain what you mean for issue 1? I thought it is an asterisk configuration issue, specially many articles in the internet have spoken about SRV, but they did not mention how to solve it.

the method I use to follow during configuration of vicidial is as follows.

http://www.eflo.net/VICIDIALforum/viewtopic.php?t=8924&sid=b2ccf1f121263c6ef0c3b165a10fb2cb 

here is the vicidial setup manual

All we use to do is make our computer talk with internet .

Configure Trunks

Configure extensions

register eyebeam

asterisk -r > show sip registry ( to check if the computer is connected to sip trunk)

load leads and start calling.

try using Trixbox if you and new to asterisk. its a gui based solution of asterisk.
trainings are also available over internet .

Unfortunately I have some requirements to use the standard asterisk instead of others.
I have used the command:
asterisk -r > sip show registry
and I got the following:
sip show registry
Host                           dnsmgr Username       Refresh State                Reg.Time                
dynamic:5060                   N      6000               105 Registered           Mon, 15 Feb 2010 08:30:02
voipbuster.xx:5060           N      xxxxxx           105 Registered           Mon, 15 Feb 2010 08:29:47
sipgate.xx:5060                N      xxxxxx           105 Registered           Mon, 15 Feb 2010 08:28:49
voipbuster.xx:5060           N      xxxxxx           105 Registered           Mon, 15 Feb 2010 08:29:47
sipgate.xx:5060               N       xxxxxx           105 Registered           Mon, 15 Feb 2010 08:28:49
5 SIP registrations.
[Feb 15 08:31:47] WARNING[3064]: acl.c:385 ast_get_ip_or_srv: Unable to lookup 'dynamic'
[Feb 15 08:31:47] WARNING[3064]: acl.c:385 ast_get_ip_or_srv: Unable to lookup 'dynamic


Why I can see 2 registration for each of sip trunks?
The problem "unable to lookup 'dynamic' is that same message I am getting /var/log/asterisk/messages, how can I solve it?

When I ran the following command:
sip show peers
Name/username                 Host            Dyn   Nat ACL Port     Status    
2000/2000                             127.0.0.1        D     N      5060     Unmonitored
my-codecs                            (Unspecified)                 5060     Unmonitored
sip_proxy                               127.0.0.1                        5060     Unmonitored
sipgate-out/xxxxxxx            xxx.xxx.xxx.xxx      N      5060     OK (41 ms)
trunk_1/xxxxxxx                 xxx.xxx.xxx.xxx               5060     Unmonitored
trunk_2/xxxxxxx                 xxx.xxx.xxx.xxx               5060     Unmonitored
ulaw-phone                           (Unspecified)                 5060     Unmonitored
voipbuster-out/xxxxxxx       xxx.xxx.xxx.xxx     N       5060     OK (39 ms)
8 sip peers [Monitored: 2 online, 0 offline Unmonitored: 6 online, 0 offline]
I can see sipgate has 2 enteries, and voipbuster has 2 entries as well, where one of them as trunk, and the other as sipgate-out, or voipbuster-out, is that correct? or I need to modify something to have only single entry?
I can see sipgate-out, and voipbuster-out are defined in sip.conf, and trunk_1, and trunk_2 are only defined in extension.conf, what shall I do?

For issue 1: I have found that host = dynamic is defined in my users.conf for my extension 600, do I need to change it to something else to remove the annoying warning  messages?
For issue 7: I have also found trunk_1, and trunk_2 are defined in users.conf, do we need to define sipgate and voipbusters in both users.conf, and sip.conf?

Well you are correct, may be I did that just because I am fed up from the problems I had in Asterisk, and and the time I spent in it.
I have spent more time and I managed to solve many issues:
1) I am always getting in /var/log/asterisk/messages the following error:
      WARNING[3064] acl.c: Unable to lookup 'dynamic'

Solution: Because the sip user is defined in users.conf by the Asterisk GUI, and that because of the Asterisk GUI.
               Definie users in sip.conf, and comment them in the users.conf solve the problem.

      2) I am getting in /var/log/asterisk/messages :
     app_system.c: Unable to execute 'dahdi_scan > /etc/asterisk/dahdi_scan.conf
     Although I do not have any dahdi devices.

Solution:  Commenting out the DAHDI/G2 in extensions.conf solve the problem.

3) I am getting in /var/log/asterisk/messages:
 WARNING[4127] http.c: fwrite() failed: Broken pipe

Solution: That is because the Asterisk formal GUI (relaese 2.0), it is bad do not use it.

4) I am getting in the /var/log/asterisk/messages:
 NOTICE[13412] res_config_ldap.c: Cannot load LDAP RealTime driver.
 WARNING[2945] res_config_ldap.c: No directory user found, anonymous binding as default.
    Although I do not have any LDAP.

No Solution Yet, but it happens only during the startup.

5) I can see the server is listening to tcp port 5060, and not UDP:
    From netstat -an:
     tcp        0      0 0.0.0.0:5060                0.0.0.0:*                   LISTEN
    udp        0      0 0.0.0.0:5060                0.0.0.0:*                              
    I have read in sip.conf I can use tcpbindaddress, udpbindaddress. or bindaddress, currently I am using bindaddress as the first 2 options did not help.

      No Solution Yet.

   6) I managed to connect to the PBX from my lan, but I am getting this error when I am trying to make a call:
  NOTICE[3064] chan_sip.c: Call from '6000' to extension '001xxxxxxxxx' rejected because extension not found, although I have created a calling rule in my extensions.conf.

Solution:  I need a stun server, I have installed the opal package, but I am not sure how to start it up, I will open a new question for that.

   7) In my sipgate account I can find 2 devices connected and not one, and both using my useragent name.

Solution: Again Because the sip GWY is defined in users.conf by the Asterisk GUI, and I define it manually in sip.conf.
               Define SIP GWY  in sip.conf, and comment them in the users.conf solve the problem, and is much better.


   8) The Asterisk GUI is very slow when it startup to check the folder rights, I have changed many file permissions to 777 for the asterisk, but it did not help.

Solution: Because the index.js in /var/lib/asterisk/statc-http/config/js is looking for user [tmp_managerUser] defining a dummy user will decrease the delay, but the GUI by itself is very bad.

    9) The Asterisk GUI can not list the voice messages, in the internet it says I have to use the listfiles script for previous releases, I found in the diguim site only release 1, and 2, I used the listfiles for release 1, and it did not solve the problem.

      No Solution, as that is because of a bug in /var/lib/asterisk/static-http/config/js/sysinfo.js where function getsysinfohtml is not defined, and no solution for it.

Thanks for all who helped me.