FritzBox im internen Netz an Asterisk anmelden

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rev0lutio

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So, nach tagelangem suchen und probieren hoffe ich mal das mir hier jemand helfen kann.

Ich habe eine FritzBox 7050 mit FW 14.04.30 welche über die LAN Schnittstelle am * angemeldet werden soll.

Die Sip.conf am * sieht so aus:
Code: 
[general]
context = default
port = 5060
bindaddr = 0.0.0.0
srvlookup = yes                  
nat = yes
externip = meine.externe.ip

[studio2]
type = friend
secret = ******
nat = no
host = dynamic
context = studio2
qualify = yes
insecure = very

Der Fritzbox verpasse ich Benutzernamen, PW und die interne IP für Registrar und Proxy (hier 192.168.10.15 für den *)

Folgendes passiert jetzt auf dem Asterisk:
Code: 
Apr 16 23:13:14 NOTICE[7699]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 16 23:13:14 WARNING[7699]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 16 23:13:14 WARNING[7699]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms

Diese Meldung kommt jetzt so ca. 10-15 mal in Abständen von ca. 4 sek.
Danach kommt dann endlich ein
Code: 
    -- Registered SIP 'studio2' at 169.254.1.1 port 5060 expires 1800
    -- Saved useragent "AVM FRITZ!Box Fon WLAN 7050 14.04.33 (May 10 2007)" for peer studio2

Beim Testen der Verbindung mittels Anruf von aussen über eine IAX Verbindung: Fehlanzeige
Ein "sip debug peer studio2" bringt
Code: 
SIP Debugging Enabled for IP: 169.254.1.1:5060
Da erscheint mir schonmal die vom * ermittelte IP zweifelhaft
Desweiteren erwartet mich bei der Debugging Ausgabe folgendes:
Code: 
    -- Accepting AUTHENTICATED call from 83.125.8.46, requested format = 2, actual format = 2
    -- Executing Ringing("IAX2/[email protected]:4569/2", "") in new stack
    -- Executing Wait("IAX2/[email protected]:4569/2", "3") in new stack
    -- Executing Dial("IAX2/[email protected]:4569/2", "Sip/studio2") in new stack
We're at 79.216.170.60 port 10018
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[email protected];uniq=B5DD099EF0D4B3B079660D809D8BD SIP/2.0
Via: SIP/2.0/UDP 79.216.170.60:5060;branch=z9hG4bK7b3e9147
From: "Radio B46 Redaktion" <sip:[email protected]>;tag=as375b1b14
To: <sip:[email protected];uniq=B5DD099EF0D4B3B079660D809D8BD>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 16 Apr 2009 21:23:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 7699 7699 IN IP4 79.216.170.60
s=session
c=IN IP4 79.216.170.60
t=0 0
m=audio 10018 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 169.254.1.1:5060
    -- Called studio2
    -- SIP/studio2-3991 is ringing
und siehe da, trotz "nat = no", Asterisk behauptet ja selbst "(no NAT)", wird der Fritzbox, die als "studio2" registriert ist, ein "We're at 79.216.170.60 port 10018" übermittelt. Hierbei handelt es sich um meine ext. IP.
Was geht hier vor?

Schalte ich nun auf nat=yes registriert sich die FritzBox per sofort und der Sprachkanal wird korrekt aufgebaut. Allerdings dann über die externe IP.
Also, wie bekomme ich es nun hin das eine korrekt Registrierung per sofort auch bei nat=no erfolgt und die interne IP übergeben wird.

Theoretisch könnte ich das auch über die externe IP laufen lassen aber hierzu sei gesagt das das ganze später innerhalb eines VPN Tunnels laufen soll und der Sprechkanal ja dann außerhalb des Tunnels aufgebaut werden würde was wiederum heißt das ich an der Firewall des Routers hinter dem der * hängt diverse Ports freigeben muss was nicht gewünscht ist.

Ich nehme mal an das das Problem von der Fritzbox her kommt. Da ein GXP-2000 die gewünschten Resultate erziehlt.

Was kann ich tun?
 
So lange du beim Asterisk für die 7050 169.254.1.1:5060 siehst obwohl LAN A der Box 192.168.10.xyz hat kann das nie und nimmer was werden.

Wie ist LAN A der 7050 eingerichtet, NAT oder IP-Client?

rev0lutio schrieb:
Der Fritzbox verpasse ich Benutzernamen, PW und die interne IP für Registrar und Proxy (hier 192.168.10.15 für den *)
Zum Vergrößern anklicken....
Den Eintrag für Proxy löschen, brauchst du nicht.

Du musst von der Box erst einmal folgendes sehen

Code: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.xyz:5060;branch=
From: <sip:[email protected]>;tag=
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 3 REGISTER
Contact: <sip:[email protected];uniq=
Authorization: Digest username="studio2", realm="asterisk", nonce="", uri="sip:192.168.10.15", response="", algorithm=MD5
Expires: 1800
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.33 (May 10 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0
darauf dann ein SIP/2.0 100 Trying und SIP/2.0 200 OK vom Asterisk
 
@stinkstiefel:
Aufgrund der ausführungen im anderen Thread gelingt es mir nach herunterfahren dar lan:0 schnittstelle und neustart des voipd in der box das die 169.254.1.1 verschwindet und 192.168.10.109 verwendet wird, was aber leider auch nichts an der Tatsache ändert das * trotz nat=no seine externe ip für die rtp ports an die box schickt und auch am mehrfachen versuch einer registrierung ändert sich nichts.
btw, wie lege ich den in der box fest das lan:0 nach Neustart der Box nicht wieder hoch fährt? Ich würds ja in die debug.cfg eintragen, die lässt sich nur leider nicht bearbeiten.
 
Vielleicht versuchst du mal auf die noch offenen Fragen zu antworten.
Hast du die 7050 schon auf Werkseinstellungen zurückgesetzt?
Wie ist LAN A der 7050 eingerichtet, NAT oder IP-Client?
Wurde der Eintrag für Proxy inzwischen gelöscht?
 
Achso, sorry, die Kiste bringt mich echt zum verzeweifeln.
Auf Werkseinstellung zurückgesetzt: Ja, mehrfach.
Lan A: IP Cient
Eintrag für Proxy Client: gelöscht
 
APIPA muss man auch ohne einen Hack los werden. Wurde die Box in irgend einer Form modifiziert? Lass mal ein Recover drüberlaufen und teste dann noch einmal.
 
Recover werd ich dann später mal probieren.
So nebenbei würde mich mal interessieren ob jemand anders eine 7050 mit FW14.04.33 in einem internen Netz erfolgreich ohne NAT an einem Asterisk am laufen hat.
 
So, hier mal die neusten Erkenntnisse:

Recover durchgeführt und Box dabei auf FW 14.04.31 downgegraded.
Box von DSL auf Internetzugang im LAN mitbenutzen umgestellt.
Box IP auf lokale Bedürfnisse angepasst.
Internettelefonie eingerichtet.
Erfolgreich! Die APIPA is weg. Mir deucht da gerade ganz dunkel irgendwo mal was gelesen zu haben das diese APIPA Sache ein Bug in der 14.04.33 sein soll. Bin mir aber grade nicht ganz sicher.

Also, Verbindung steht. Nach wie vor braucht die Box aber ca. 10 Versuche um vom Asterisk registriert zu werden.
Hier mal der log der Anmeldeprozedur der FritzBox
Code: 
asterisk*CLI> 

Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bKF8DA1BEF43682735
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


14 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bKF8DA1BEF43682735
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bKF8DA1BEF43682735
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
WWW-Authenticate: Digest realm="asterisk", nonce="031cf378"
Content-Length: 0


 to 192.168.10.109:5060
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:20 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:20 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:20 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:21 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:21 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:21 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:25 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:25 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:25 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:29 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:29 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:29 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:33 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:33 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:33 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:37 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:37 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:37 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:41 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:41 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:41 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:45 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:45 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:45 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


15 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0B903B962AD2A896
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Apr 17 21:29:49 NOTICE[6675]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:29:49 WARNING[6675]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:29:49 WARNING[6675]: chan_sip.c:4914 register_verify: Failed to parse contact info
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
REGISTER sip:192.168.10.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0E870935BB051F26
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 3 REGISTER
Contact: <sip:[email protected];uniq=B5DD099EF0D4B3B079660D809D8BD>
Authorization: Digest username="studio2", realm="asterisk", nonce="031cf378", uri="sip:192.168.10.15", response="25a8f57d5344a8ee7835d846a8d38e5c"
Expires: 1800
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


17 headers, 0 lines
Using latest request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0E870935BB051F26
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bK0E870935BB051F26
From: <sip:[email protected]>;tag=2362944985
To: <sip:[email protected]>;tag=as72c7262c
Call-ID: [email protected]
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 1800
Contact: <sip:[email protected];uniq=B5DD099EF0D4B3B079660D809D8BD>;expires=1800
Date: Fri, 17 Apr 2009 19:29:52 GMT
Content-Length: 0


 to 192.168.10.109:5060
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read: 
SUBSCRIBE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bKEEAFDB4CD2239D38
From: <sip:[email protected]>;tag=956239502
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 4 SUBSCRIBE
Contact: <sip:[email protected];uniq=B5DD099EF0D4B3B079660D809D8BD>
Event: message-summary
Expires: 3600
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb  5 2007)
Allow: NOTIFY
Accept: application/simple-message-summary
Content-Length: 0


14 headers, 0 lines
Using latest SUBSCRIBE request as basis request
Sending to 192.168.10.109 : 5060 (non-NAT)
Found peer 'studio2'
Looking for studio2 in studio2
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.109:5060;branch=z9hG4bKEEAFDB4CD2239D38
From: <sip:[email protected]>;tag=956239502
To: <sip:[email protected]>;tag=as6bfc2310
Call-ID: [email protected]
CSeq: 4 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 192.168.10.109:5060
Destroying call '[email protected]'

so siehts ohne Sip Debug aus:
Code: 
Apr 17 21:43:11 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:11 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:11 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:11 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:11 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:11 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:15 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:15 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:15 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:19 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:19 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:19 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:23 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:23 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:23 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:27 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:27 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:27 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:31 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:31 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:31 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:35 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:35 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:35 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
Apr 17 21:43:39 NOTICE[7449]: chan_sip.c:4489 parse_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway
Apr 17 21:43:39 WARNING[7449]: chan_sip.c:4518 parse_contact: Invalid host ''
Apr 17 21:43:39 WARNING[7449]: chan_sip.c:4914 register_verify: Failed to parse contact info
    -- Saved useragent "AVM FRITZ!Box Fon WLAN 7050 14.04.31 (Feb 5 2007)" for peer studio2
 
Hmmm,....ok.
Allerdings tritt dieser Fehler nicht auf wenn ich die Box von aussen durchs NAT connecten lasse.
 
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