Problem: Asterisk ignoriert INVITE Messages

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stani2

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Hallo!

Wir experimentieren in der Schule mit Asterisk und SIP.
Bis jetzt hatten wir dabei auch keine großen Probleme. Jedoch ignoriert Asterisk seit kurzem die von uns gesendeten INVITE-Messages. Und leider konnten wir den Fehler bisher nicht finden.

Bei uns läuft Asterisk v1.4 auf einem ubuntu System.

Ich hab hier mal den SIP-debug. Vielleicht kann uns dadurch jemand weiter helfen.

Zuerst mal der REGISTER-Teil:
Code: 
<--- SIP read from 192.168.100.17:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:[email protected]:5070>
Expires: 3600
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.17 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>;tag=as0d8811c3
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12105d19"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.100.17:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:[email protected]:5070>;q=;expires=
Expires: 3600
Content-Length: 0
Authorization: Digest username="2000", realm="asterisk", nonce="12105d19", uri="192.168.100.32", response="474418e865cc40a226cd6d54820f0bce", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.17 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
    -- Saved useragent "TwinSIP Communicator" for peer 2000

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17
From: <sip:[email protected]>;tag=16cD
To: <sip:[email protected]>;tag=as0d8811c3
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:[email protected]:5070>;expires=3600
Date: Thu, 24 Apr 2008 15:29:15 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '[email protected]' Method: REGISTER

<--- SIP read from 192.168.100.50:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:[email protected]:5070>
Expires: 3600
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.50 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO;received=192.168.100.50
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO;received=192.168.100.50
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>;tag=as77afec68
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06c7a8b6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'U$Ö[email protected]' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.100.50:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport=
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:[email protected]:5070>;q=;expires=
Expires: 3600
Content-Length: 0
Authorization: Digest username="2001", realm="asterisk", nonce="06c7a8b6", uri="192.168.100.32", response="0097b6ddf1cec8fc636ccb938dcb0e38", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.50 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport=;received=192.168.100.50
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
    -- Saved useragent "TwinSIP Communicator" for peer 2001

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport=;received=192.168.100.50
From: <sip:[email protected]>;tag=YilY
To: <sip:[email protected]>;tag=as77afec68
Call-ID: U$Ö[email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:[email protected]:5070>;expires=3600
Date: Thu, 24 Apr 2008 15:29:51 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'U$Ö[email protected]' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'U$Ö[email protected]' Method: REGISTER

Nun der INVITE-Teil:
Hier funktioniert das ganze leider nicht mehr.
Code: 
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99

v=0
o=andy 2634 2634 IN IP4 192.168.100.50
s=test von alex
t=0 0
m=audio 9050 udp PCM/32000/8/1

<------------->
--- (10 headers 5 lines) ---
Sending to 192.168.100.50 : 5070 (no NAT)
Using INVITE request as basis request - [email protected]

<--- Reliably Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
Found user '2001'

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #1 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #2 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #3 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #4 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #5 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
Retransmitting #6 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7"
Content-Length: 0


---

<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:[email protected]>
To: <sip:[email protected]>;tag=as3887b859
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
[SIZE="4"][COLOR="Red"][B]Ignoring this INVITE request[/B][/COLOR][/SIZE]
[Apr 24 17:31:57] WARNING[5391]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission [email protected] for seqno 3 (Non-critical Response)
Really destroying SIP dialog '[email protected]' Method: INVITE

Nun würden wir gerne wissen, was die Ursache dafür sein kann. Haben wir vielleicht einen Fehler in den SIP-Nachrichten?
Wir wären für jede Hilfe Dankbar.

Mfg.
Alex

Edit Guard-X: Code-Tags eingefügt!
 
Die CSeq-Nummer wird nicht hochgezählt.
 
@FrankIT:

Danke für deine Antwort.
Ich habe es nun so gemacht, dass bei jeder Nachricht die CSeq-Nummer um 1 inkrementiert wird.
Nun erhalte ich vom Server eine 491 Request Pending-Message und danach wieder die Challange usw.
 
Ich habe den Fehler schon gefunden.
Die Ursache war, dass ich bei der ACK-Nachricht auch die CSeq-Nummer erhöht habe und asterisk somit diese anscheinend nicht mit der passenden Invite-Nachricht assoziieren konnte.

Nochmals Danke für die Hilfe.
 
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