[gelöst] Nach update geth nix mehr

[snoopy_spy]

Hallo, nach einem update funktioniert jetzt nix mehr, mein snom sagt mir immer "Not acceptable here"

Hier ist der SIP log vom telefon

Sent to udp:192.168.11.13:5060 at 27/8/2009 16:24:11:591 (1180 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-yp1v6jw0mv6c;rport
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2048>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 755178922 755178922 IN IP4 192.168.11.14
s=call
c=IN IP4 192.168.11.14
t=0 0
m=audio 57034 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:5H57l/P14wGxfVTj1Qxa4GLyi66u7QYeMoiFEFK1
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

Received from udp:192.168.11.13:5060 at 27/8/2009 16:24:11:607 (523 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-yp1v6jw0mv6c;received=192.168.11.14;rport=2048
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>;tag=as69c90134
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f2f7aa7"
Content-Length: 0

Sent to udp:192.168.11.13:5060 at 27/8/2009 16:24:11:612 (376 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-yp1v6jw0mv6c;rport
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>;tag=as69c90134
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2048>;flow-id=1
Content-Length: 0

Sent to udp:192.168.11.13:5060 at 27/8/2009 16:24:11:623 (1362 bytes):

INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-ka7wc7u9nl1y;rport
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2048>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="15",realm="asterisk",nonce="7f2f7aa7",uri="sip:[email protected];user=phone",response="bd0c8cba83e2b25744d675a77c80e835",algorithm=MD5
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 755178922 755178922 IN IP4 192.168.11.14
s=call
c=IN IP4 192.168.11.14
t=0 0
m=audio 57034 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:5H57l/P14wGxfVTj1Qxa4GLyi66u7QYeMoiFEFK1
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

Received from udp:192.168.11.13:5060 at 27/8/2009 16:24:11:680 (435 bytes):

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-ka7wc7u9nl1y;received=192.168.11.14;rport=2048
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>;tag=as69c90134
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Sent to udp:192.168.11.13:5060 at 27/8/2009 16:24:11:686 (376 bytes):

ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.14:2048;branch=z9hG4bK-ka7wc7u9nl1y;rport
From: "andi" <sip:[email protected]>;tag=knxvu701wh
To: <sip:[email protected];user=phone>;tag=as69c90134
Call-ID: 3c6cafa47978-c42only392ww
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2048>;flow-id=1
Content-Length: 0

die sip conf dür diesen aparat

[15]
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=1
pickupgroup=1
dial=SIP/15
accountcode=
mailbox=15@default
permit=192.168.1.1/255.255.0.0
callerid=device <15>
call-limit=50

 

[snoopy_spy]

... gelöst, es waren auch die codecs ...