My First Asterisk CLI MESSAGES

Hello,

First of all i’m from Germany . Sorry for my English.

I have a Debian 7 with Asterisk 1.8 . Everythink is fine, but on the CLI i get Messages that I’dont understand.
Could you explain / help me?!

[code]Reliably Transmitting (NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 177.7.7.55:5060;branch=z9hG4bK12837d22;rport
Max-Forwards: 70
From: “asterisk” sip:XXXXXXX@177.7.7.55;tag=as2e8772db
To: sip:sipgate.de
Contact: sip:XXXXXXX@177.7.7.55:5060
Call-ID: 6ab3c8cf38340de27ae76fc24c277f20@177.7.7.55:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0
Date: Tue, 27 Aug 2013 13:18:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:217.10.79.9:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 177.7.7.55:5060;branch=z9hG4bK12837d22;rport=54772;received="EXTERNAL_IP"
From: “asterisk” sip:XXXXXXX@177.7.7.55;tag=as2e8772db
To: sip:sipgate.de;tag=64c295986a77a1f756ad49f3e6513d0d.20a2
Call-ID: 6ab3c8cf38340de27ae76fc24c277f20@177.7.7.55:5060
CSeq: 102 OPTIONS
Accept: /
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘6ab3c8cf38340de27ae76fc24c277f20@177.7.7.55:5060’ Method: OPTIONS

<— SIP read from UDP:217.10.79.9:5060 —>

<------------->
[Aug 27 15:18:19] NOTICE[1216]: chan_sip.c:13652 sip_reregister: – Re-registration for XXXXXXX@sipgate.de
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 217.10.79.9:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 177.7.7.55:5060;branch=z9hG4bK2e2041cb;rport
Max-Forwards: 70
From: sip:XXXXXXX@sipgate.de;tag=as322d3e83
To: sip:XXXXXXX@sipgate.de
Call-ID: 3815d0bb2b3f47c07e51985d768cccff@127.0.1.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.23.0
Authorization: Digest username=“XXXXXXX”, realm=“sipgate.de”, algorithm=MD5, uri=“sip:sipgate.de”, nonce=“521ca900ea891ae9810745085b4b70959aefb31b”, response="a7fd958e7ddeaff9a66bfcc58c4d1a63"
Expires: 120
Contact: sip:XXXXXXX@177.7.7.55:5060
Content-Length: 0


<— SIP read from UDP:217.10.79.9:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 177.7.7.55:5060;received=“EXTERNAL_IP”;branch=z9hG4bK2e2041cb;rport=54772
From: sip:XXXXXXX@sipgate.de;tag=as322d3e83
To: sip:XXXXXXX@sipgate.de;tag=fbf1d80521ea9f98078b6998e7669f9b.ad67
Call-ID: 3815d0bb2b3f47c07e51985d768cccff@127.0.1.1
CSeq: 104 REGISTER
Contact: sip:XXXXXXX@177.7.7.55:5060;expires=120;received="sip:“EXTERNAL_IP”:54772"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Scheduling destruction of SIP dialog ‘3815d0bb2b3f47c07e51985d768cccff@127.0.1.1’ in 32000 ms (Method: REGISTER)
[Aug 27 15:18:29] NOTICE[1216]: chan_sip.c:21525 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)

<— SIP read from UDP:217.10.79.9:5060 —>[/code]

Here my sip.conf

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=de

;Ansteuerung LED
;allowsubscribe = yes
;notifyringing = yes
;notifyhold = yes
;limitonpeers = yes
;---------------------------------------
;Registrieung des Asterisk bei Sipgate
register => XXXXX:XXXXXX@sipgate.de/XXXXXX

[sip-out]
type=friend
insecure=invite
username=XXXXXX
fromuser=XXXXXX
fromdomain=sipgate.de
secret=XXXXXX
host=sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=ulaw,alaw,g729,gsm,slinear
context=ankommend[/code]

here my extensions.conf

[code][general]
static=yes
writeprotect=no

[lokal]
;Erreichbarkeit der Nebenstellen 30-3X
;untereinander, nach 20 sek. mailbox
;exten => 30,hint,SIP/30
;exten => 31,hint,SIP/31
;exten => 32,hint,SIP/32

exten => _3X,1,Dial(SIP/${EXTEN},20)
exten => _3X,n,VoiceMail(${EXTEN},u)

;exten => 40,1,VoiceMail(40,b)

[abgehend]
;----------------------------
;abgehende Anrufe über Sip-Account (sip-out)

;exten => _X.,1,Set(CALLERID(num)=XXXXX)
exten => _X.,1,Dial(SIP/${EXTEN}@sip-out,30,trg)
;exten => _X.,3,Hangup

[ankommend]
;Anrufe auf der externen Adrese klingeln auf der 30,
;31 und 32. wer zuerst abnimmt hat gewonnen. Nach 15
;Sekunden wird auf die Mailbox bei Nr. 30 gesporchen
;gruppenmailbox einrichten.

exten => 2359314,1,Dial(SIP/30&SIP/31&SIP/32,15)
exten => 2359314,n,Goto(r-${DIALSTATUS},1)

;Wenn besetzt
exten => r-BUSY,1,voicemail(40,b)
exten => r-BUSY,2,hangup()

;Wenn keine Antwort
exten => r-NOANSWER,1,voicemail(40,u)
exten => r-NOANSWER,2,hangup()

[eigene_mailbox]
exten => 88,1,answer()
exten => 88,n,wait(1)
exten => 88,n,VoiceMailMain(s${CALLERID(num)})
exten => 88,n,hangup()

[mailbox]
exten => 80,1,answer()
exten => 80,n,wait(1)
exten => 80,n,voicemailmain()
exten => 80,n,hangup()

[default]
include => lokal
include => eigene_mailbox
include => mailbox
include => abgehend

Hi,

Those are SIP trace messages, you can turn those off by running “sip set debug off” through Asterisk CLI

testpbx*CLI> sip set debug off SIP Debugging Disabled

Thank you for your answer.

This is no Error or something else?

greetz from germany

On a quick look, there doesn’t seem to be an error, but, if you think there is an error, you need to tell us the symptoms, as looking for problems in SIP traces is quite time consuming, even when one has some idea of what one is looking for.