Asterisk chan_sip.c:12219 sip_reg_timeout:

Asterisk Asterisk Support
1

Hi guys!
i try to configure my asterisk so that from mysophone i can dial a PSTN line through My sipgate provider
but i receive this eror from asterisk.

please can you tell me what wrong ?

doing dnsmgr_lookup for ‘sipgate.de’
ast_get_srv: SRV lookup for ‘_sip._udp.sipgate.de’ mapped to host sipgate.de, port 5060
[Jan 12 15:45:33] NOTICE[29103]: chan_sip.c:12219 sip_reg_timeout: – Registration for 'XXXXXXX@sipgate.de’ timed out, trying again (Attempt #140)

my sip.conf

[general]
port=5060
bindaddr=0.0.0.0
context= sonstige

register =>XXXXXXX:YYYYYYY@sipgate.de/XXXXXXX

[1000]
type=friend
context=meine-telefone
secret=1234
host=dynamic

[1001]
type=friend
context=meine-telefone
secret=1234
host=dynamic

[1002]
type=friend
secret=1234
host=dynamic

[1003]
type=friend
secret=1234
host=dynamic

[1004]
type=friend
secret=1234
host=dynamic

[1005]
type=friend
secret=1234
host=dynamic

[ext-sip-account] ;nach außen telefonnieren

type=friend
context=von-sipgate
username =XXXXXXX
fromuser =XXXXXXX
secret =YYYYYYY
host=sipgate.de
fromdomain=sipgate.de
quality=yes
insecure=port,invite
nat=yes

my extenxions.conf

[default]

exten => 2000,1,Answer() ; abgehoben
exten => 2000,2,Playback(hello-world) ;playback Helloword
exten => 2000,3,Hangup() ; aufgelegt

[sonstige]

[meine-telefone]

exten => 1000,1,Dial(SIP/1000,20) ;nach 20 second geht die anruf auf die Voicemail
exten => 1000,2,VoiceMail(1000,u) ;The letter u, if present, causes the unavailable message to be played. ;By default, the message says, “The person at extension … 1000 … is ; unavailable,”

                                   ; The letter b, if present in voicemail parameter, causes the busy
                                   ; message to be played.
                                   ; By default, the message says, "The person at extension ... 1000
                                   ; ... is busy." 

                                   ; The letter s, if present, causes the instructions 
                                   ;("Please leave your message after the tone. When done, hang up, or
                                   ; press the pound key.") to be skipped.

exten => 1001,1,Dial(SIP/1001)
exten => 1001,2,VoiceMail(1001,u)

exten => 1002,1,Dial(SIP/1002,20)
exten => 1002,2,VoiceMail(1002,u)

exten => 1003,1,Dial(SIP/1003)
exten => 1003,2,VoiceMail(1003,u)

exten => 1004,1,Dial(SIP/1004,20)
exten => 1004,2,VoiceMail(1004,u)

exten => 1005,1,Dial(SIP/1005)
exten => 1005,2,VoiceMail(1005,u)

exten => 999,1,VoiceMailMain(${CALLERID(num)},s) ;the SIP user to geth his voicemail message have to call
; the 999 with her owne phone to hear her messages
; when s ist as parameter they Skip checking the passcode
; for the mailbox.

thank U

I hope you can help me

Im junior in Asterisk i just make my test.

exten => _0[1-9].,1,Dial(SIP/${EXTEN}@sipgate) ; context zum außen Telefonnieren mit SIPGATE