Trunk not Register

In this server I have a SIP trunk where he logs on the server and typically makes connections (a month), already the register (to receive calls) does not work in reality but when you have already set up a ‘sip show registry’ I returned empty, I tried all it does not register in any way, my files, settings below.

SIP.CONF

register => 9000:xxxxxxxx@sip.mydomain.com.br:5060/9000

[9000]
type=friend
username=9000
secret=xxxxxxxx
fromuser=9000
fromdomain=sip.mydomain.com.br
context=entrada
disallow=all
allow=g729
host=sip.mydomain.com.br
insecure=very
qualify=yes
port=5060
nat=yes
dtmfmode=rfc2833

CONSOLE

Connected to Asterisk 1.4.22 currently running on asterisk (pid = 24616)
Verbosity is at least 3

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
9000/9000                      myip      N      5060     OK (50 ms)

asterisk*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time           
asterisk*CLI>

A connection being made…

    -- Executing [01881188380@saida:1] Dial("SIP/20-0f8783e0", "SIP/01881188380@9000|60|TthHC")                                                               in new stack
    -- Called 01881188380@9000
    -- SIP/9000-0f879ca0 is making progress passing it to SIP/20-0f8783e0
asterisk*CLI>

Any help is welcome :smile:

Do you have issues with just incoming ? Did you try running ngrep (network grep) to see if anything is coming in to the server on an incoming call ? Do you have an extension 9000 in context entrada?

Yes, only at the entrance!, Calls placed normally, even you can see, that not enough to register, if any errors show was already a step forward, but that is how it does not happen anything.

I appreciate your help, that day is posted in the forum because it really gave up seeking a solution on the Internet, I do not think anywhere, the principle is fine …

asterisk*CLI> core show uptime
System uptime: 1 day, 16 hours, 45 minutes, 17 seconds
asterisk*CLI>
asterisk*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
asterisk*CLI>

Context entrada:

[entrada]

include => manha|00:00-12:00|mon-sun|*|*
include => tarde|12:01-18:00|mon-sun|*|*
include => noite|18:01-23:59|mon-sun|*|*

Only complement it, I have a FXO gateway that is in the context entrada and operates normally

Ok. The issue is that in the context entrada you do not have an extension 900 there. You need to tell asterisk what to do when a call comes in to 9000 (as you have in your register statement. Also you have

include => manha|00:00-12:00|mon-sun||
include => tarde|12:01-18:00|mon-sun||
include => noite|18:01-23:59|mon-sun||

Which is completely wrong. You are trying to include Gotoiftime paramaters as a context. Please post your entire extensions.conf

; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your 
; inbound and outbound calls in Asterisk. 
; 
; This configuration file is reloaded 
; - With the "dialplan reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.  
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens. 
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command "dialplan save" too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of 
; things to do, Asterisk will wait for a new extension to be dialed 
; (this is the original behavior of Asterisk 1.0 and earlier).
;
;autofallthrough=no
;
; If clearglobalvars is set, global variables will be cleared 
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a "reload" will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with "reload" in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
;priorityjumping=yes
;
; User context is where entries from users.conf are registered.  The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp				; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest					; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/G2					; Trunk interface
;
; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
;    (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;    (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
;    time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;    time (aka. descending rotary hunt group).
;
TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider

;
; Any category other than "General" and "Globals" represent 
; extension contexts, which are collections of extensions.  
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches 
;	anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;       it can unambiguously determine that no other matches are possible
;
; For example the extension _NXXXXXX would match normal 7 digit dialings, 
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.  The priority
; "next" or "n" means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension.  Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').  
; Priorities may then also have an alias, or label, in 
; parenthesis after their name which can be used in goto situations
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. 
;
;[context]
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;exten => someexten,{priority|label{+|-}offset}[(alias)],application,arg1|arg2...
;
; Included Contexts
;
; One may include another context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
; The reason a context would include other contexts is for their 
; extensions.
; The algorithm to find an extension is recursive, and works in this
; fashion: 
;	 first, given a stack on which to store context references, 
;           push the context to find the extension onto the stack...
;    a) Try to find a matching extension in the context at the top of 
;       the stack, and, if found, begin executing the priorities
;       there in sequence.
;    b) If not found, Search the switches, if any declared, in
;       sequence.
;    c) If still not found, for each include, push that context onto 
;       the top of the context stack, and recurse to a).
;    d) If still not found, pop the entry from the top of the stack; 
;       if the stack is empty, the search has failed. If it's not, 
;       continue with the next context in c).
; This is a depth-first traversal, and stops with the first context 
; that provides a matching extension. As usual, if more than one
; pattern in a context will match, the 'best' match will win.
; Please note that that extensions found in an included context are
; treated as if they were in the context from which the search began.
; The PBX's notion of the "current context" is not changed.
; Please note that in a context, it does not matter where an include
; directive occurs. Whether at the top, or near the bottom, the effect 
; will be the same. The only thing that matters is that if there is 
; more than one include directive, they will be searched for extensions 
; in order, first to last.
; Also please note that pattern matches (like _9XX) are not treated
; any differently than exact matches (like 987). Also note that the
; order of extensions in a context have no affect on the outcome.
;
; Timing list for includes is 
;
;   <time range>|<days of week>|<days of month>|<months>
;
; Note that ranges may be specified to wrap around the ends.  Also, minutes are
; fine-grained only down to the closest even minute.
;
;include => daytime|9:00-17:00|mon-fri|*|*
;include => weekend|*|sat-sun|*|*
;include => weeknights|17:02-8:58|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/G2/${EXTEN:7}) ; Expose all of 256-428 
;exten => _1256325XXXX,1,Dial(Zap/G2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using 
; the Local channel driver. 
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases, 
; you have to goto "s" to execute that extension.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote 
; IAX switching you transparently get access to the remote
; Asterisk PBX
; 
; switch => IAX2/user:password@bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}

[macro-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should 
; terminate call)
;   ${ARG1} - What to dial
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20)			; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

[macro-stdPrivacyexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p)			; Ring the interface, 20 seconds maximum, call screening 
						; option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1)		; Callee chose to send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)		; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

[macro-page];
;
; Paging macro:
;
;       Check to see if SIP device is in use and DO NOT PAGE if they are
;
;   ${ARG1} - Device to page

exten => s,1,ChanIsAvail(${ARG1}|js)			; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup


[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait(1)			; Wait a second, just for fun
exten => s,n,Answer			; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
exten => s,n,WaitExten			; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
exten => 3,n,Goto(s,restart)		; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip)		; "Please hold while..." 
					; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})

exten => 1235,1,Voicemail(1234,u)		; Right to voicemail

exten => 1236,1,Dial(Console/dsp)		; Ring forever
exten => 1236,n,Voicemail(1234,b)		; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
exten => #,n,Hangup			; Hang them up.

;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)			; If they take too long, give up
exten => i,1,Playback(invalid)		; "That's not valid, try again"

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)		; Return to the start over message.

;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
exten => 600,n,Echo			; Do the echo test
exten => 600,n,Playback(demo-echodone)	; Let them know it's over
exten => 600,n,Goto(s,6)		; Start over

;
;	You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)

; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;
;	The page context calls up the page macro that sets variables needed for auto-answer
;	It is in is own context to make calling it from the Page() application as simple as 
;	Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
;
; By default we include the demo.  In a production system, you 
; probably don't want to have the demo there.
;
include => demo

;
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)

; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict.  You can alias
; them with names, too, and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT)	; Use hint as listed
;exten => 6245,n,Voicemail(6245,u)		; Voicemail (unavailable)
;exten => 6245,s+1,Hangup			; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b)	; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}

;exten => 6275,1,Macro(stdexten,6275,${MARK})	; assuming ${MARK} is something like Zap/2
;exten => mark,1,Goto(6275|1)			; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL})	; Ditto for wil
;exten => wil,1,Goto(6236|1)

;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
;
; "core show application <command>" will show details of how you
; use that particular application in this file, the dial plan. 
; "core show functions" will list all dialplan functions
; "core show function <COMMAND>" will show you more information about
; one function. Remember that function names are UPPER CASE.

[entrada]

include => manha|00:00-12:00|mon-sun|*|*
include => tarde|12:01-18:00|mon-sun|*|*
include => noite|18:01-23:59|mon-sun|*|*

[manha]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Playback(sejabemvindo)
exten => s,4,Playback(bom dia)
exten => s,5,Playback(aguarde)
exten => s,6,Playback(gravada)
exten => s,7,Wait(2)
exten => s,8,Goto(suporte,s,1)

[tarde]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(HORIGEM=${CALLERID(all)})
exten => s,n,Set(CALLFILENAME=${DNID}/${STRFTIME(${EPOCH},,%d-%m-%Y+%H-%M)}.${HORIGEM}.)
exten => s,n,Monitor(wav,/monitor/${CALLFILENAME},m)
exten => s,n,Playback(sejabemvindo)
exten => s,n,Playback(boa tarde)
exten => s,n,Playback(aguarde)
exten => s,n,Playback(gravada)
exten => s,n,Wait(2)
exten => s,n,Goto(suporte,s,1)

[noite]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(sejabemvindo)
exten => s,n,Playback(boa noite)
exten => s,n,Playback(aguarde)
exten => s,n,Playback(gravada)
exten => s,n,Goto(suporte,s,1)

[suporte]
exten => s,1,Dial(SIP/20,35|m)
exten => s,2,Dial(SIP/mytel@9000,15,r)
exten => s,3,Playback(indisponivel)
exten => s,4,Congestion()
exten => s,5,Hangup()

[saida]

exten => 1000,1,Goto(entrada,s,1) ; to test

exten => _0,1,Dial(SIP/${EXTEN}@41,60,TthHC)

exten => _XX,1,SIPAddHeader(Alert-Info: Bellcore-r2)
exten => _XX,2,Macro(SIP/${EXTEN},35)

exten => _X.,1,Dial(SIP/${EXTEN}@9000,60,TthHC)
exten => _0.,1,Dial(SIP/${EXTEN}@9000,60,TthHC)

Try changing your entrada context to:

[entrada]
exten => 9000,1,GotoIfTime(00:00-12:00|sun-sat|*|*?manha,s,1)
exten => 9000,2,GotoIfTime(12:01-18:00|sun-sat|*|*?tarde,s,1)
exten => 9000,3,GotoIfTime(18:01-23:59|sun-sat|*|*?noite,s,1)

You also may want to read up a bit on how the includes and dial plan formatting in general.

It did not work, could free up remote access for you to have a look what do you think?

The Asterisk should not register first and then give some kind of error in extensions.conf?

or alternative?

You would likely see an error in /var/log/asterisk/full and not in asterisk its self. If you would like me to remote in PM me.