Hello, everybody, I am connected to a server that is in another city very far away from me, the things that I want that aah does, it does, but there isnt sound, I cant hear the ring, the messages that I play with playback command, I cant hear them, WHY???, in the console i can see that all my commands are executed including “Hangup” but my sip phone stills connected, what can I do to fix this problem?, please help me…, In advance thanks 
well, you can help us help you by providing a LOT more information.
what type of phone, what codecs, what is your bandwidth, etc…post some output from the CLI…
as baconbuttie is fond of saying, we cannot read minds.
just what i was thinking 
Im using the xlite, I am connected to 100 mbps bandwidth, and I dont know what codecs are installed in the server, how can I know this information, please help
well, do you manage the server or does someone else? if you do, you need to check your sip.conf file to see what codecs you have enabled.
past that, are you or the asterisk server behind a router or firewall? if so, you will need to set up port forwarding so that RTP traffic can get through and be routed correctly.
I am not control the server, but this information from sip.conf
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
Other thing, when I dial a wrong extension, I can hear the message saying me “the person is unavalible please…” but WHY I can hear this message and mines not? and why my phone stills connected after the hangup?, I hope that information from sip. conf be useful
that’s not what i need - you need the actual sip.conf entry for YOUR phone.
that looks like the info from the general section.
you might also get a hold of whoever manages the server and get the output from the CLI when the call is attempted - if this is a codec issue, then the CLI should indicate so.
it is my sip.conf entry for my xlite phone
[1234]
username=1234
type=friend
secret=test
;record_out=On-Demand
;record_in=On-Demand
qualify=yes
port=5060
nat=yes
mailbox=1234@default
;host=dynamic
;dtmfmode=rfc2833
;context=from-internal
context=prueba
;canreinvite=no
;callerid=“test” <1234>
this my context prueba
[prueba]
exten => 1234,1,Answer
exten => 1234,2,Playback(demo-thanks)
exten => 1234,3,Playback(welcome)
exten => 1234,4,Background(year)
exten => 1234,5,SayDigits(1234)
exten => 1234,6,Hangup
and I can see all messages from asterisk server, cuz I am using putty to connect me to it, all intructions are executed, but I cant hear the messages and the most important THING, when hangup is executed my phone stills connected, why???
describe the network layout - you have nat=yes and qualify=yes, but say that you’re on a 100Mbit network - is the asterisk server on the same network or a different one? is there a NAT in between you and the server?