Configure Asterisk

I am a computer science student. I am learning about asterisk. I installed goautodial ce-2.1-final. It contains: Asterisk
My newly installed system is unable to make outgoing calls.
Here is the configurations I think necessary. Please tell me what changes do I make?

My career configuration:=

Carrier ID: 	santanu
blah blah blah.....
Registration String: register => *********:*********
Account Entry:

fromuser=[same as username]
trustrpid = yes
sendrpid = yes
canreinvite = yes

Protocol: SIP

exten => _91XXXXXXXXXX,1,AGI(agi://
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@tintoring,,tTo)
exten => _91XXXXXXXXXX,3,Hangup 
My sip.conf

context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=                ; IP address to bind to ( binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld            ; Set default domain for this host
;pedantic=yes                   ; Enable checking of tags in headers,
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
language=en                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
videosupport=no                 ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes                  ; generate manager events when sip ua
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
;rtptimeout=60                   ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
rtpkeepalive=60            ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;recordhistory=yes              ; Record SIP history by default
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
notifyringing = yes             ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes                ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes              ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes            ; Default false
;register =>
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
;externip =        ; Address that we're going to put in outbound SIP
;     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
localnet=; All RFC 1918 addresses are local networks
localnet=     ; Also RFC1918
localnet=          ; Another RFC1918 with CIDR notation
localnet= ;Zero conf local network
nat=yes                         ; Global NAT settings  (Affects all peers and users)
canreinvite=no          ; Asterisk by default tries to redirect the
;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes              ; Save systemname in realtime database at registration
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
;domain=                 ; Add IP address as local domain
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld        ; When making outbound SIP INVITEs to
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100             ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes             ; By default, qualify all peers at 2000ms
limitonpeer = yes       ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=
; dtmfmode=inband
; qualify=1000
Realtime Logs:

-- Executing [8600051@default:1] MeetMe("Local/8600051@default-87b6,2", "8600051|F") in new stack
   > Channel Local/8600051@default-87b6,1 was answered.
 -- Executing [914167466946@default:1] AGI("Local/8600051@default-87b6,1", "agi://") in new stack
 -- AGI Script agi:// completed, returning 0
  -- Executing [914167466946@default:2] Dial("Local/8600051@default-87b6,1", "SIP/14167466946@tintoring||tTo") in new stack
 == Parsing '/etc/asterisk/manager.conf': [Jan 24 23:47:18] VERBOSE[12056] logger.c: [Jan 24 23:47:18] Found
 == Manager 'sendcron' logged on from
 -- Executing [58600051@default:1] MeetMe("Local/58600051@default-1f04,2", "8600051|Fmq") in new stack
  > Channel Local/58600051@default-1f04,1 was answered.

I just noticed
sip show peers ->
tintoring/7824454609 N 5060 UNREACHABLE
may be this is the cause. How to resolve it?

Unreachable means that it failed to respond to OPTIONS request, not even with a rejection. The fix is to fix whatever is causing it not respond. This could be many things, in many places.

In the relatively unlikely event that it isn’t responding because it has buggy handing of unknown methods, you could try qualify=no, which turns off this test.

If it is responding, but extremely late, you will probably need to fix this but it is possible to set qualify to an even longer time.

Incidentally, why are you starting to learn on a version of Asterisk that is past its maintenance end of life?