Asterisk -> Asterisk | 1Account OK | 2Accounts FEHLER

thisismyname

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Guten morgen liebe vieltelefonierer,

mal wieder ein Problem der ganz besonderen Sorte. Bei mir steht links ein Asterisk und rechts ein Asterisk. Links soll sich auf Rechts mit sip accounts anmelden. Mit einem Account geht das wunderbar. Das ganze schaut dann so aus:

Code:
EXTENSION.CONF
[general]
port=5060
bindaddr=0.0.0.0
srvlookup = yes
defaultexpiry=600

;register => 3000:[email protected]/3000
register => 3001:[email protected]/3001

;[3000]
;context=tel3000
;type=peer
;secret=1234
;busy-level=1000
;username=3000
;fromuser=3000
;secret=1234
;host=10.0.0.11
;Asterisk sip auth = 3000

[3001]
context=tel3001
type=friend
secret=1234
busy-level=1000
username=3001
fromuser=3001
secret=5678
host=10.0.0.11
Asterisk sip auth = 3001
Und hier der SIP TRACE:

Code:
*CLI> REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK534105a1;rport
From: <sip:[email protected]>;tag=as74bf8857
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---

<--- SIP read from 10.0.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK534105a1;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as74bf8857
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from 10.0.0.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK534105a1;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as74bf8857
To: <sip:[email protected]>;tag=as703af09d
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2489e1b5"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name 10.0.0.11
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK759841a2;rport
From: <sip:[email protected]>;tag=as231fda3e
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="3001", realm="asterisk", algorithm=MD5, uri="sip:10.0.0.11", nonce="2489e1b5", response="0b6558e9e17d9bb17669482f17514d79", opaque=""
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---

<--- SIP read from 10.0.0.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK759841a2;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as231fda3e
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from 10.0.0.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK759841a2;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as231fda3e
To: <sip:[email protected]>;tag=as703af09d
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 600
Contact: <sip:[email protected]>;expires=600
Date: Tue, 21 Oct 2008 07:20:11 GMT
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
Will ich das ganze nun mit 2 Accounts machen... funktioniert gar nichts mehr. Die sip.conf:
Code:
[general]
port=5060
bindaddr=0.0.0.0
srvlookup = yes
defaultexpiry=600

register => 3000:[email protected]/3000
register => 3001:[email protected]/3001

[3000]
context=tel3000
type=peer
secret=1234
busy-level=1000
username=3000
fromuser=3000
secret=1234
host=10.0.0.11
Asterisk sip auth = 3000

[3001]
context=tel3001
type=friend
secret=1234
busy-level=1000
username=3001
fromuser=3001
secret=5678
host=10.0.0.11
Asterisk sip auth = 3001
Und der SIP TRACE:

Code:
*CLI> REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK4f933bdb;rport
From: <sip:[email protected]>;tag=as65d78cc8
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
[Oct 21 09:33:15] NOTICE[13669]: chan_sip.c:7400 sip_reregister:    -- Re-registration for  [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK0e1d230f;rport
From: <sip:[email protected]>;tag=as0f3dc45f
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
Retransmitting #1 (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK4f933bdb;rport
From: <sip:[email protected]>;tag=as65d78cc8
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
[Oct 21 09:33:55] NOTICE[13669]: chan_sip.c:7430 sip_reg_timeout:    -- Registration for '[email protected]' timed out    , trying again (Attempt #1)
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK3f700a7b;rport
From: <sip:[email protected]>;tag=as13402696
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
Retransmitting #1 (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK0e1d230f;rport
From: <sip:[email protected]>;tag=as0f3dc45f
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
[Oct 21 09:34:35] NOTICE[13669]: chan_sip.c:7430 sip_reg_timeout:    -- Registration for '[email protected]' timed out, trying again (Attempt #1)
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.11:5060:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6e32d258;rport
From: <sip:[email protected]>;tag=as017834fa
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0
Das seltsame ist, verfolge ich das ganze mit wireshark bekomme ich sehrwohl eine Antwort... die schaut dann so aus:

Code:
REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>;tag=as7ab4c199
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f6657ab"
Content-Length: 0

REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK01bbd603;rport
From: <sip:[email protected]>;tag=as0844cc26
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

REGISTER sip:10.0.0.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK01bbd603;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as0844cc26
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK01bbd603;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as0844cc26
To: <sip:[email protected]>;tag=as1ddf11ef
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6ccca2c9"
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK6022e64a;rport;received=10.0.0.10
From: <sip:[email protected]>;tag=as65357d71
To: <sip:[email protected]>;tag=as53a9cd02
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e5e17b4"
Content-Length: 0
Ich hab schon sehr viel mit type=friend/peer herrumgespielt und auch schon einige andere Sachen versucht... Ich denke im Prinzip sollte mal nur eine Option aendern muessen und dann muesste es gehen... nur welche? ;)

Ich hoff irgnedwer von euch kann mir helfen... would be nice, greetz

myname
 

doxon

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Mal ne generelle Frage,

wozu brauchst du die mehreren Accounts denn?
Evtl. Ist eine Lösung mit IAX ja schöner.
 

Burmann

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Mit
secret=1234
secret=5678

bei einem SIP-Account funktioniert wohl nix.

Aber abgesehen davon wird's auch nicht gehen, da Asterisk (ankommend) die Accounts nicht auseinanderhalten kann. Dies bedeutet wenn bei allen das PW gleich ist, wird's gehen. Welchen Account er dann ankommend benutzt wird muß man ausprobieren. Abgehender Ruf wird der richtige (den man im Dial angegeben hat) genommen.
 

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