[Problem] 1und1 Nummern extern nicht erreichbar

Harsesis

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Hallo zusammen,

wie in meinem ersten Thread in diesem Forum (im sipgate Unterforum) geschrieben habe ich probleme damit Asterisk mit 1und1 zum laufen zu bringen. Ich habe mir hier im Forum bereits einen Wolf gesucht aber noch keine Lösung gefunden. Vereinzelt bin ich ebenso auf darauf gestoßen das das ganze wohl garnicht einwandfrei möglich sei? Eigentlich hatte ich daher 1und1 aufgegeben und wollte einen Umweg über sipgate bauen, jedoch habe ich es einfach noch einmal versucht und siehe da es ging kurzezeit. Ich verwende Asterisk 10.1.2 mit dieser sip.conf:

Code:
[gerneral]
bindport = 5060
bindaddr = 0.0.0.0
srvlookup=yes
externhost=*****.dyndns.org
localnet=192.168.1.0/255.255.255.0
nat=yes
register => 4922*******8:*******@sipgate.de/4922*******8
register => 4922*******7:*******@sipgate.de/4922*******7

[2000]
type=friend
secret=1234
host=dynamic
context=meine-telefone

[2001]
type=friend
secret=1234
host=dynamic
context=meine-telefon


[1und1](!)
type=friend
disallow=all
allow=alaw
allow=ulaw
;allow=g729
allow=g726
allow=gsm
host=sip.1und1.de
fromdomain=1und1.de
qualify=yes
insecure=port,invite
tos=0x18
context=incoming
;caninvite=no
canreinvite=no
dtmfmode=auto
language=de

[1und1-1-1](1und1)
host=sipbalance1-1.1und1.de

[1und1-1-2](1und1)
host=sipbalance1-2.1und1.de

[1und1-2-1](1und1)
host=sipbalance2-1.1und1.de
[1und1-2-2](1und1)
host=sipbalance2-2.1und1.de

[1und1-3-1](1und1)
host=sipbalance3-1.1und1.de

[1und1-3-2](1und1)
host=sipbalance3-2.1und1.de

[1und1-4-1](1und1)
host=sipbalance4-1.1und1.de

[1und1-4-2](1und1)
host=sipbalance4-2.1und1.de

[1und1-5-1](1und1)
host=sipbalance5-1.1und1.de

[1und1-5-2](1und1)
host=sipbalance5-2.1und1.de

[1und1-6-1](1und1)
host=sipbalance6-1.1und1.de

[1und1-6-2](1und1)
host=sipbalance6-2.1und1.de

[1und1-7-1](1und1)
host=sipbalance7-1.1und1.de

[1und1-7-2](1und1)
host=sipbalance7-2.1und1.de

[1und1-8-1](1und1)
host=sipbalance8-1.1und1.de

[1und1-8-2](1und1)
host=sipbalance8-2.1und1.de

[telefonica-1](1und1)
host=1und1-1.sip.mgc.voip.telefonica.de

[telefonica-2](1und1)
host=1und1-2.sip.mgc.voip.telefonica.de

[telefonica-3](1und1)
host=1und1-3.sip.mgc.voip.telefonica.de

[telefonica-4](1und1)
host=1und1-4.sip.mgc.voip.telefonica.de

[telefonica-5](1und1)
host=1und1-5.sip.mgc.voip.telefonica.de

[telefonica-6](1und1)
host=1und1-6.sip.mgc.voip.telefonica.de

[telefonica-7](1und1)
host=1und1-7.sip.mgc.voip.telefonica.de

[telefonica-8](1und1)
host=1und1-8.sip.mgc.voip.telefonica.de

[4922*********8](1und1)
context=incomming
username=4922********8
secret=********
fromuser=4922**********8
host=sip.1und1.de

[4922********7](1und1)
context=incomming
username=4922*********7
secret=*********
fromuser=4922*********7
host=sip.1und1.de
sowie diese extensions.conf:

Code:
[default]

[meine-telefone]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()

exten => 2000,1,Dial(SIP/2000,10)
exten => 2000,2,VoiceMail(2000, u)

exten => 2001,1,Dial(SIP/2001)
exten => 2001,2,VoiceMail(2001,u)

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)

exten => _0[1-9].,1,Dial(SIP/${EXTEN}@4922*******8)

[incoming]
exten => 4922********8,1,Dial(SIP/2001)
exten => 4922********7,1,Dial(SIP/2000)
Anrufe nach Außen klappen problemlos. Eingehende gingen erst nicht, dann nach einem weiteren Versuch ging es für villeicht eine Stunde und nun geht es nicht mehr. Ich habe in der Zeit nichts an der Configuration geändert. Ich bekomme in der CLI (verbosity 5) keinen Eingehenden Anruf angezeigt. Ebenso sind keine Events in der Firewall zu sehen.

Evtl. habt ihr ja noch einen Tipp.
 

abw1oim

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Wenn die sip.conf richtig wiedergegeben ist, sind die 1und1-Nummern nicht bei 1und1 registriert, sondern es wird der Versuch unternommen, sie bei sipgate zu registrieren:

register => 4922*******8:*******@sipgate.de/4922*******8
register => 4922*******7:*******@sipgate.de/4922*******7
Das kann nicht funktionieren!

Ansonsten:
Bei 1und1 besteht das grundsätzliche Problem, dass die Registrierungsintervalle extrem lang sind, was sich mit dynamischen IP-Adressen leider nicht allzu gut verträgt (Stichwort: IP-Wechsel bei Verbindungstrennung). Also entweder eine Verbindungstrennung erkennen und mit einem sip reload reagieren oder aber die 1und1-Nummern auf einem Host mit statischer IP (vServer oder ähnliches) auflaufen lassen.
 

Harsesis

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Hi,

danke für deine Antwort! Mit dem Register hast du natürlich recht, habe ich direkt gändert, muss wohl ein Übertragungsfehler gewesen sein bei dem ganzen rumprobieren. Hat jedoch auch nichts geändert (Ich frage mich wie das so überhaupt klappen konnte?). Müsst ein "sip show registry" nicht ebenfalls etwas ausgeben? Die liste ist bei mir immer leer.

Bzg. der Langen Intervalle: Meine IP Ändert sich natürlich täglich (Privatanschluss) jedoch immer zu einem Festen Zeitpunkt. Bietet Asterisk eine möglichkeit entweder das die registrierung in regelmäßigen kurzen intervallen zu aktualisieren oder jeweils zu einem festen Zeitpunkt?
 

abw1oim

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Bietet Asterisk eine möglichkeit entweder das die registrierung in regelmäßigen kurzen intervallen zu aktualisieren oder jeweils zu einem festen Zeitpunkt?
Ja - allerdings macht da 1und1 nicht mit. Da hilft nur ein sip reload, das man ggf. über einen cron-Job ansteuern kann, so man denn die Disconnect-Zeit hat und sich da "dranklemmt".

Und ja: sip show registry sollte etwas zeigen. Ist dies nicht der Fall, kann ggf. sip set debug on weiterhelfen (ggf. den Output hier posten), da dann ersichtlich wird, welche SIP-Pakete wohin gehen und was ggf. für Antworten eintreffen.
 

Harsesis

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Hi,

ich habe den debug eingeschaltet und danach einen reload gemacht, dies war die ausgabe:

Code:
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.138:5060:
OPTIONS sip:sipbalance8-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1d80ba7f;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as0c8caef7
To: <sip:sipbalance8-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.205:5060:
OPTIONS sip:sipbalance5-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK0bef9c19;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as16593f0f
To: <sip:sipbalance5-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.138:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1d80ba7f;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as0c8caef7
To: <sip:sipbalance8-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.53e5
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.205:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK0bef9c19;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as16593f0f
To: <sip:sipbalance5-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a12e
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.135:5060:
OPTIONS sip:sipbalance5-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2edb9b84;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as20005bef
To: <sip:sipbalance5-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.204:5060:
OPTIONS sip:sipbalance4-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3335bf41;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as69028d1b
To: <sip:sipbalance4-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2edb9b84;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as20005bef
To: <sip:sipbalance5-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7715
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '795e82fa5382be3c40a4829[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.204:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3335bf41;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as69028d1b
To: <sip:sipbalance4-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.bab6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.134:5060:
OPTIONS sip:sipbalance4-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4646d585;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as712b3130
To: <sip:sipbalance4-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.137:5060:
OPTIONS sip:sipbalance7-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1e346b00;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as00bd4f0c
To: <sip:sipbalance7-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4646d585;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as712b3130
To: <sip:sipbalance4-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.28ec
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.137:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1e346b00;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as00bd4f0c
To: <sip:sipbalance7-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0f3f
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.197:5060:
OPTIONS sip:sipbalance7-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2a6a166f;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as3912ca62
To: <sip:sipbalance7-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.136:5060:
OPTIONS sip:sipbalance6-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK00a98f87;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7af7ab44
To: <sip:sipbalance6-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.197:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2a6a166f;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as3912ca62
To: <sip:sipbalance7-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7258
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK00a98f87;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as7af7ab44
To: <sip:sipbalance6-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c33b
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.206:5060:
OPTIONS sip:sipbalance6-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1acb5067;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as55836f89
To: <sip:sipbalance6-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.206:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK1acb5067;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as55836f89
To: <sip:sipbalance6-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.4e1a
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.201:5060:
OPTIONS sip:sipbalance1-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7b8c76cd;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as4bbab4d2
To: <sip:sipbalance1-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7b8c76cd;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as4bbab4d2
To: <sip:sipbalance1-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.5c05
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.131:5060:
OPTIONS sip:sipbalance1-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2edaf2ae;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1ce0b3bc
To: <sip:sipbalance1-1.1und1.de>
Contact: <sip:[email protected]1.9:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.133:5060:
OPTIONS sip:sipbalance3-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2b579715;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as2d96ace9
To: <sip:sipbalance3-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2edaf2ae;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as1ce0b3bc
To: <sip:sipbalance1-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.45fe
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK2b579715;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as2d96ace9
To: <sip:sipbalance3-1.1und1.de>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cddf
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.203:5060:
OPTIONS sip:sipbalance3-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK46bcdde4;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1fa52aeb
To: <sip:sipbalance3-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.132:5060:
OPTIONS sip:sipbalance2-1.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6e06e8b4;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as5cb60373
To: <sip:sipbalance2-1.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.203:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK46bcdde4;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as1fa52aeb
To: <sip:sipbalance3-2.1und1.de>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-61e6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.132:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK6e06e8b4;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as5cb60373
To: <sip:sipbalance2-1.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a23d
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.202:5060:
OPTIONS sip:sipbalance2-2.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK56b75b43;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as59f1e354
To: <sip:sipbalance2-2.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 212.227.67.134:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7ad96d26;rport
Max-Forwards: 70
From: "asterisk" <sip:4922********[email protected]>;tag=as167dd2d9
To: <sip:sip.1und1.de>
Contact: <sip:4922********[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK56b75b43;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as59f1e354
To: <sip:sipbalance2-2.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f361
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:212.227.67.134:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7ad96d26;rport=5060
From: "asterisk" <sip:4922********[email protected]:5060>;tag=as167dd2d9
To: <sip:sip.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.ca12
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.227.67.136:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK0376a4c3;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7a10b125
To: <sip:sip.1und1.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 193.189.245.140:5060:
OPTIONS sip:1und1-8.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK026f3172;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as676fec76
To: <sip:1und1-8.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.227.67.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK0376a4c3;rport=5060
From: "asterisk" <sip:[email protected]:5060>;tag=as7a10b125
To: <sip:sip.1und1.de>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0624
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:193.189.245.140:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK026f3172;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as676fec76
To: <sip:1und1-8.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 193.189.245.140:5060:
OPTIONS sip:1und1-5.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK70a41cf7;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7a542c23
To: <sip:1und1-5.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:193.189.245.140:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK70a41cf7;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as7a542c23
To: <sip:1und1-5.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 193.189.245.138:5060:
OPTIONS sip:1und1-6.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK784cdc7c;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as22140340
To: <sip:1und1-6.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 193.189.245.139:5060:
OPTIONS sip:1und1-7.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3843cefa;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as188e1ba9
To: <sip:1und1-7.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:193.189.245.138:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK784cdc7c;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as22140340
To: <sip:1und1-6.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:193.189.245.139:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK3843cefa;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as188e1ba9
To: <sip:1und1-7.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 193.189.245.208:5060:
OPTIONS sip:1und1-1.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK73678203;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as77ae7c30
To: <sip:1und1-1.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 193.189.245.209:5060:
OPTIONS sip:1und1-2.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK211179f0;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as51af10ed
To: <sip:1und1-2.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:193.189.245.208:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK73678203;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as77ae7c30
To: <sip:1und1-1.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:193.189.245.209:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK211179f0;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as51af10ed
To: <sip:1und1-2.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 193.189.245.138:5060:
OPTIONS sip:1und1-3.sip.mgc.voip.telefonica.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK75a94513;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as55b90f21
To: <sip:1und1-3.sip.mgc.voip.telefonica.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 22 Feb 2012 09:59:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:193.189.245.138:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK75a94513;rport
From: "asterisk" <sip:[email protected]:5060>;tag=as55b90f21
To: <sip:1und1-3.sip.mgc.voip.telefonica.de>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
 

abw1oim

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Da gibt es kein einziges REGISTER-Paket.
Insoweit kann an der register-Anweisung in Deiner sip.conf etwas nicht stimmen, da eben gar kein Registrierungsversuch ausgelöst wird (die aufgeführten OPTIONS-Pakete sind in Ordnung, stammen aber ausschließlich von den deklarierten peers).
 

Harsesis

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Aber woran kann das liegen? Die oben geposteten .conf dateien sind 100% mit meinen Deckungsgleich (bis auf den tausch von sipgate.de durch sip.1und1.de). Ist die verwendung der Version 10.1.2 evtl. nicht zu empfehlen? Ich verstehe einfach nicht warum das nicht klappt!
 

abw1oim

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Ich würde es mal wie folgt probieren:

Code:
[gerneral]
bindport = 5060
bindaddr = 0.0.0.0
srvlookup=yes
externhost=*****.dyndns.org
localnet=192.168.1.0/255.255.255.0
nat=yes
[B];register ersetzt durch callbackextension
;register => 4922*******8:*******@sipgate.de/4922*******8
;register => 4922*******7:*******@sipgate.de/4922*******7[/B]

[2000]
type=friend
secret=1234
host=dynamic
context=meine-telefone

[2001]
type=friend
secret=1234
host=dynamic
context=meine-telefon


[1und1](!)
[B];type ersetzt durch peer
;type=friend
type=peer[/B]
[B];codecs reduziert[/B]
disallow=all
allow=alaw
allow=ulaw
;allow=g729
[B];allow=g726
;allow=gsm[/B]
host=sip.1und1.de
fromdomain=1und1.de
qualify=yes
insecure=port,invite
tos=0x18
context=incoming
;caninvite=no
[B];Syntaxkompatibilität -> canreinvite ist deprecated
;canreinvite=no
directmedia=no[/B]
dtmfmode=auto
language=de

[1und1-1-1](1und1)
host=sipbalance1-1.1und1.de

[1und1-1-2](1und1)
host=sipbalance1-2.1und1.de

[1und1-2-1](1und1)
host=sipbalance2-1.1und1.de
[1und1-2-2](1und1)
host=sipbalance2-2.1und1.de

[1und1-3-1](1und1)
host=sipbalance3-1.1und1.de

[1und1-3-2](1und1)
host=sipbalance3-2.1und1.de

[1und1-4-1](1und1)
host=sipbalance4-1.1und1.de

[1und1-4-2](1und1)
host=sipbalance4-2.1und1.de

[1und1-5-1](1und1)
host=sipbalance5-1.1und1.de

[1und1-5-2](1und1)
host=sipbalance5-2.1und1.de

[1und1-6-1](1und1)
host=sipbalance6-1.1und1.de

[1und1-6-2](1und1)
host=sipbalance6-2.1und1.de

[1und1-7-1](1und1)
host=sipbalance7-1.1und1.de

[1und1-7-2](1und1)
host=sipbalance7-2.1und1.de

[1und1-8-1](1und1)
host=sipbalance8-1.1und1.de

[1und1-8-2](1und1)
host=sipbalance8-2.1und1.de

[telefonica-1](1und1)
host=1und1-1.sip.mgc.voip.telefonica.de

[telefonica-2](1und1)
host=1und1-2.sip.mgc.voip.telefonica.de

[telefonica-3](1und1)
host=1und1-3.sip.mgc.voip.telefonica.de

[telefonica-4](1und1)
host=1und1-4.sip.mgc.voip.telefonica.de

[telefonica-5](1und1)
host=1und1-5.sip.mgc.voip.telefonica.de

[telefonica-6](1und1)
host=1und1-6.sip.mgc.voip.telefonica.de

[telefonica-7](1und1)
host=1und1-7.sip.mgc.voip.telefonica.de

[telefonica-8](1und1)
host=1und1-8.sip.mgc.voip.telefonica.de

[4922*********8](1und1)
context=incomming
[B];Syntaxkompatibilität -> username ist deprecated
;username=4922********8
defaultuser=4922********8[/B]
secret=********
fromuser=4922**********8
host=sip.1und1.de
[B];Ergänzung für Registrierung
callbackextension=4922********8 ;-> Wert hier entspricht Wert für defaultuser[/B]


[4922********7](1und1)
context=incomming
[B];Syntaxkompatibilität -> username ist deprecated
;username=4922********7
defaultuser=4922********7[/B]
secret=*********
fromuser=4922*********7
host=sip.1und1.de
[B];Ergänzung für Registrierung
callbackextension=4922********8 ;-> Wert hier entspricht Wert für defaultuser[/B]
Die fetten Markierungen dienen nur dem Auffinden der Änderungen!
 

Harsesis

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Hi,

ein RIESENGROSSES DANKE an dich, so funktioniert es!

Eine Frage zu dem peer: ich hatte irgendwo gelesen das gäbe es in den neuen Versionen nicht mehr, nurnoch user und friend? Muss wohl falsch gewesen sein...

Beim erneuten durchsuchen meines logs bin ich auf diese Zeilen gestoßen:

Code:
[Feb 22 03:11:57] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '16220181933423' rejected because extension not found in context 'default'.
[Feb 22 03:11:57] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '0442032989340' rejected because extension not found in context 'default'.
[Feb 22 03:11:59] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '00442032989350' rejected because extension not found in context 'default'.
[Feb 22 03:12:02] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '000442032989341' rejected because extension not found in context 'default'.
[Feb 22 03:12:05] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '0001442032989342' rejected because extension not found in context 'default'.
[Feb 22 03:12:08] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '001442032989346' rejected because extension not found in context 'default'.
[Feb 22 03:12:11] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '9442032989342' rejected because extension not found in context 'default'.
[Feb 22 03:12:14] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '900442032989353' rejected because extension not found in context 'default'.
[Feb 22 03:12:17] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '9011442032989345' rejected because extension not found in context 'default'.
[Feb 22 03:12:20] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '90442032989352' rejected because extension not found in context 'default'.
[Feb 22 03:12:23] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '011442032989348' rejected because extension not found in context 'default'.
[Feb 22 03:12:26] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+00442032989352' rejected because extension not found in context 'default'.
[Feb 22 03:12:29] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+011442032989353' rejected because extension not found in context 'default'.
[Feb 22 03:12:32] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+0442032989343' rejected because extension not found in context 'default'.
[Feb 22 03:12:35] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+900442032989349' rejected because extension not found in context 'default'.
[Feb 22 03:12:38] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+9011442032989340' rejected because extension not found in context 'default'.
[Feb 22 03:12:41] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '+442032989347' rejected because extension not found in context 'default'.
[Feb 22 03:12:44] NOTICE[9103] chan_sip.c: Call from '' (94.231.83.134:5060) to extension '442032989344' rejected because extension not found in context 'default'.
Hat da etwa schon jemand versucht meinen Asterisk zu "Hacken" bzw zu "Missbrauchen"?
 

abw1oim

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peer, user und friend existieren alle noch und nioch immer wird mehr oder minder philosophiert, was denn nun eigentlich genutzt werden soll ... Faktisch funktioniert peer aber eben.

Die logs zeigen tatsächlich Einbruchsversuche mit teils kreativen Zielformaten :)

Da empfiehlt sich eineseits allowguest=no im general-Teil der sip.conf (ist aber bei 1und1 kritisch, da man deren Serverfarm in der Regel nicht vollständig abbildet und dann eingehende Anrufe auf der Strecke bleiben könnten, siehe andere Threads hier im Forum).
Alternativ lies mal zu fail2ban nach, da gibt es auch hier im Forum so einiges ...
 

Harsesis

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Ah oki super zu wissen! Ich habe nun mal Fail2ban installiert, würde das ganze jedoch gerne mal testen, kannst du mir sagen wie (ob) ich den obigen Angriff einfach nachbilden kann? Ein externer Zugriff steht zur verfügung, bin mir jedoch nicht ganz sicher wie ich das ganze simulieren kann...
 

abw1oim

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Im Prinzip vom (externen) Sip-Telefon oder (externen) Asterisk mit

Code:
<rufnummer>@*****.dyndns.org
dabei ist *****.dyndns.org die dyndns-Adresse Deines (internen) Asterisk
 

Harsesis

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Ich habe nun versucht fail2ban dazu zu bringen angreifer zu blocken die auf diese Weise versuchen über mich zu telefonieren, jedoch weigert sich fail2ban irgendwie noch etwas. Ich habe folgende failregex angelegt:

Code:
failregex = Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Wrong password
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - No matching peer found
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Username/auth name mismatch
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Device does not match ACL
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - Peer is not supposed to register
            Registration from '.*' failed for '<HOST>(:[0-9]{1,5})?' - ACL error (permit/deny)
            <HOST>(:[0-9]{1,5})? failed to authenticate as '.*'$
            No registration for peer '.*' \(from <HOST>(:[0-9]{1,5})?\)
            Host <HOST>(:[0-9]{1,5})? failed MD5 authentication for '.*' (.*)
            Failed to authenticate user .*@<HOST>(:[0-9]{1,5})?).*
            Call from '' (<HOST>(:[0-9]{1,5})?) to extension '.*' rejected because extension not found in context '.*'
Die ersten zeilen sind hierbei aus dem Netz, die letze Regel habe ich hinzugefügt. Jedoch scheint diese (oder womöglich alle?!) nicht zu greifen. Ich kann weiterhin Angriffe von außen simulieren, jedoch reagiert fail2ban nichta auf die events. Fail2ban läuft (zu sehen an den IPTables).

Evtl. jemand eine idee?

Edit: Habe ebenso gerade versucht allowguest=no zu setzen um dies mal zu testen. Eigenartiger weise bekomme ich trotzdem weiterhin beim externen Zugriff auf den Server den gleichen Notice, also "NOTICE[2850]: chan_sip.c:23022 handle_request_invite: Call from '' (***:55984) to extension '01779660343' rejected because extension not found in context 'default'.". Das sieht für mich nicht danach aus, dass die Zugriffe generell geblocked werden...
 
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3CX

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