Anruf wird nicht korrekt beendet

CrazyPlaya

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Hi an alle,

ich habe folgendes Szenario bei einem Session Auf- und Abbau mit einem über den Asterix Server registrierten Client.

Hier einmal mein Auszug aus Packetyzer:
Code:
INVITE sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKa4573acdd
Max-Forwards: 70
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060
Call-ID: a4573acdd27aa13c
CSeq: 6880 INVITE
Contact: <sip:[email protected]:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 225

v=0
o=664 92665721 0 IN IP4 172.16.2.161
s=Media
c=IN IP4 172.16.2.161
m=audio 4000 RTP/AVP 0 8 3 4
a=rtcp:4001 IN IP4 172.16.2.161
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000

--------------------------------------------------------------------------------------------------------

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKa4573acdd;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060;tag=as2aae458f
Call-ID: a4573acdd27aa13c
CSeq: 6880 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4badb0ad"
Content-Length: 0

--------------------------------------------------------------------------------------------------------

ACK sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKa4573acdd
Max-Forwards: 70
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060;tag=as2aae458f
Call-ID: a4573acdd27aa13c
CSeq: 6880 ACK
Content-Length: 0

--------------------------------------------------------------------------------------------------------

INVITE sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKa4573acdd
Max-Forwards: 70
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060
Call-ID: a4573acdd27aa13c
CSeq: 6881 INVITE
Contact: <sip:[email protected]:5060>
Proxy-Authorization: Digest username="664", nonce="4badb0ad", realm="asterisk", uri="sip:665@Adresse", response="1cb20942b8affdc7d7e5658295a440cd", algorithm=MD5
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 225

v=0
o=664 92665721 0 IN IP4 172.16.2.161
s=Media
c=IN IP4 172.16.2.161
m=audio 4000 RTP/AVP 0 8 3 4
a=rtcp:4001 IN IP4 172.16.2.161
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000

--------------------------------------------------------------------------------------------------------

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKa4573acdd;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060
Call-ID: a4573acdd27aa13c
CSeq: 6881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

--------------------------------------------------------------------------------------------------------

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKa4573acdd;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060;tag=as2f9f4344
Call-ID: a4573acdd27aa13c
CSeq: 6881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

--------------------------------------------------------------------------------------------------------

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKa4573acdd;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060;tag=as2f9f4344
Call-ID: a4573acdd27aa13c
CSeq: 6881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 182

v=0
o=root 5782 5782 IN IP4 172.16.2.157
s=session
c=IN IP4 172.16.2.157
t=0 0
m=audio 12012 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--------------------------------------------------------------------------------------------------------

ACK sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKa4573acdd
Max-Forwards: 70
From: sip:664@Adresse;tag=a4573acdd
To: sip:665@Adresse:5060;tag=as2f9f4344
Call-ID: a4573acdd27aa13c
CSeq: 6881 ACK
Content-Length: 0

--------------------------------------------------------------------------------------------------------

BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.2.157:5060;branch=z9hG4bK29f01a88;rport
From: sip:665@Adresse:5060;tag=as2f9f4344
To: sip:664@Adresse;tag=a4573acdd
Call-ID: a4573acdd27aa13c
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--------------------------------------------------------------------------------------------------------

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.2.157:;rport=;received=172.16.2.157;branch=z9hG4bK29f01a88
From: sip:665@Adresse:5060;tag=as2f9f4344
To: sip:664@Adresse;tag=a4573acdd
Call-ID: a4573acdd27aa13c
CSeq: 102 BYE
Content-Length: 0

--------------------------------------------------------------------------------------------------------

INVITE sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKecff74d49
Max-Forwards: 70
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060
Call-ID: ecff74d497c14321
CSeq: 2652 INVITE
Contact: <sip:[email protected]:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 225

v=0
o=664 75454563 0 IN IP4 172.16.2.161
s=Media
c=IN IP4 172.16.2.161
m=audio 4000 RTP/AVP 0 8 3 4
a=rtcp:4001 IN IP4 172.16.2.161
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000

--------------------------------------------------------------------------------------------------------

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKecff74d49;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060;tag=as1fc7f9bc
Call-ID: ecff74d497c14321
CSeq: 2652 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17d3f2e9"
Content-Length: 0

--------------------------------------------------------------------------------------------------------

ACK sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKecff74d49
Max-Forwards: 70
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060;tag=as1fc7f9bc
Call-ID: ecff74d497c14321
CSeq: 2652 ACK
Content-Length: 0

--------------------------------------------------------------------------------------------------------

INVITE sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKecff74d49
Max-Forwards: 70
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060
Call-ID: ecff74d497c14321
CSeq: 2653 INVITE
Contact: <sip:[email protected]:5060>
Proxy-Authorization: Digest username="664", nonce="17d3f2e9", realm="asterisk", uri="sip:665@Adresse", response="3e8a6653765b2ce01bc5bc562fdbcc6d", algorithm=MD5
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 225

v=0
o=664 75454563 0 IN IP4 172.16.2.161
s=Media
c=IN IP4 172.16.2.161
m=audio 4000 RTP/AVP 0 8 3 4
a=rtcp:4001 IN IP4 172.16.2.161
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000

--------------------------------------------------------------------------------------------------------

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKecff74d49;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060
Call-ID: ecff74d497c14321
CSeq: 2653 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

--------------------------------------------------------------------------------------------------------

SIP/2.0 603 Declined
Via: SIP/2.0/UDP 172.16.2.161:5060;branch=z9hG4bKecff74d49;received=172.16.2.161;rport=5060
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060;tag=as5ea55594
Call-ID: ecff74d497c14321
CSeq: 2653 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

--------------------------------------------------------------------------------------------------------

ACK sip:665@Adresse SIP/2.0
Via: SIP/2.0/UDP 172.16.2.161:5060;rport;branch=z9hG4bKecff74d49
Max-Forwards: 70
From: sip:664@Adresse;tag=ecff74d49
To: sip:665@Adresse:5060;tag=as5ea55594
Call-ID: ecff74d497c14321
CSeq: 2653 ACK
Content-Length: 0

Obwohl wenn ich von dem Protokoll ausgehe alles korrekt scheint seht ihr, das nach dem 2ten Versuch einen neue Session aufzubauen ich ein Declined zurückbekomme, welches ich mir nicht erklären kann.

Vllt. findet ja jemand einen Fehler der mir nicht aufgefallen ist.

Gruß

Karsten
 
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