[Voip] Probleme mit SNOM 360 VoIP Phone

[Stefan Schultheis (home)]

Hallo alle,

Ich verwende unter der DW 1434 das Funkfeuer-VOIP-Service seit einigen
Wochen.

Mit Android-Clients hab' ich kein Problem, das klappt.

Jetzt hat mir ein Freund ein Snom 360 zur Verfügung gestellt. Das hat auch
vor 2-3 Monaten noch funktioniert, dann plötzlich gar nicht mehr und heute
hab' ich es resettet und wollte es neu konfigurieren:

+) einbuchen kann ich mich,
+) eingehende Anrufe kommen an und funktionieren,
-) ausgehend kann ich ENUM Testnummern bzw. andere Funkfeuer-Nebenstellen
NICHT erreichen. Die Meldung ist "Not acceptable here".

Ich habe einen Trace gemacht (siehe beiliegendes .pcap). Dort fällt mir
eine "401 Unauthorized" sowie "488 Not acceptable here" (siehe auch
snom-sip.txt, das ist die reine SIP Kommunikation) auf.

Wahrscheinlich eine ganz einfache Sache, aber ich komm' nicht drauf, woran
es liegen kann...

Gem. Internet-Tipps habe ich mehrere Codecs ausprobiert, mit SRTP usw.
herumgespielt, aber immer das gleiche Ergebnis.

SW beim Snom ist aktuell (snom360-SIP 8.7.3.25.5). Der NAT Router ist ein
Routerboard750 mit RouterOS 6. STUN Server hab' ich am Snom Phone
eingetragen.

Wäre für sachdienliche Hinweise dankbar,
LgS
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INVITE sip:(spam-protected);user=phone SIP/2.0
Via: SIP/2.0/UDP 78.41.118.138:2049;branch=z9hG4bK-h1k798ekb8ej;rport
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:(spam-protected):2049;line=pqjgirdk>;reg-id=1
X-Serialnumber: 000413290DA8
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.7.3.25.5
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 531

v=0
o=root 2010892412 2010892412 IN IP4 192.168.112.133
s=call
c=IN IP4 192.168.112.133
t=0 0
m=audio 51982 TCP/RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:jDiUdvuP3xEHNOzhtTT4EysbbpFg17KeaH+kNH/w
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=connection:new
a=setup:active
a=sendrecv
NOTIFY sip:(spam-protected):2049;line=pqjgirdk SIP/2.0
Via: SIP/2.0/UDP 193.238.157.10:5060;branch=z9hG4bK3e5b434a;rport
Max-Forwards: 70
From: <sip:*99%(spam-protected)>;tag=as30b57cac
To: <sip:(spam-protected):2049;line=pqjgirdk>
Contact: <sip:*99%(spam-protected):5060>
Call-ID: (spam-protected):5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.5.1
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:(spam-protected)
Voice-Message: 0/0 (0/0)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.112.133:2049;branch=z9hG4bK-h1k798ekb8ej;received=192.168.112.133;rport=2049
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>;tag=as06405fd1
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 1 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="voip.funkfeuer.at", nonce="2c99b9c6"
Content-Length: 0

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 193.238.157.10:5060;branch=z9hG4bK3e5b434a;rport=5060
From: <sip:*99%(spam-protected)>;tag=as30b57cac
To: <sip:(spam-protected):2049;line=pqjgirdk>
Call-ID: (spam-protected):5060
CSeq: 102 NOTIFY
Content-Length: 0

ACK sip:(spam-protected);user=phone SIP/2.0
Via: SIP/2.0/UDP 78.41.118.138:2049;branch=z9hG4bK-h1k798ekb8ej;rport
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>;tag=as06405fd1
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:(spam-protected):2049;line=pqjgirdk>;reg-id=1
Content-Length: 0

INVITE sip:(spam-protected);user=phone SIP/2.0
Via: SIP/2.0/UDP 78.41.118.138:2049;branch=z9hG4bK-jiso48gttnbf;rport
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:(spam-protected):2049;line=pqjgirdk>;reg-id=1
X-Serialnumber: 000413290DA8
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.7.3.25.5
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username="stefanschultheis_762",realm="voip.funkfeuer.at",nonce="2c99b9c6",uri="sip:(spam-protected);user=phone",response="86ff460fbae7fb85dd6af811ff351b88",algorithm=MD5
Content-Type: application/sdp
Content-Length: 531

v=0
o=root 2010892412 2010892412 IN IP4 192.168.112.133
s=call
c=IN IP4 192.168.112.133
t=0 0
m=audio 51982 TCP/RTP/AVP 9 0 8 3 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:jDiUdvuP3xEHNOzhtTT4EysbbpFg17KeaH+kNH/w
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=connection:new
a=setup:active
a=sendrecv
NOTIFY sip:(spam-protected):2049;line=pqjgirdk SIP/2.0
Via: SIP/2.0/UDP 193.238.157.10:5060;branch=z9hG4bK540f682d;rport
Max-Forwards: 70
From: <sip:*99%(spam-protected)>;tag=as1daa69fc
To: <sip:(spam-protected):2049;line=pqjgirdk>
Contact: <sip:*99%(spam-protected):5060>
Call-ID: (spam-protected):5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.5.1
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:(spam-protected)
Voice-Message: 0/0 (0/0)
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.112.133:2049;branch=z9hG4bK-jiso48gttnbf;received=192.168.112.133;rport=2049
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>;tag=as06405fd1
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 193.238.157.10:5060;branch=z9hG4bK540f682d;rport=5060
From: <sip:*99%(spam-protected)>;tag=as1daa69fc
To: <sip:(spam-protected):2049;line=pqjgirdk>
Call-ID: (spam-protected):5060
CSeq: 102 NOTIFY
Content-Length: 0

ACK sip:(spam-protected);user=phone SIP/2.0
Via: SIP/2.0/UDP 78.41.118.138:2049;branch=z9hG4bK-jiso48gttnbf;rport
From: <sip:(spam-protected)>;tag=xf1o2kxltz
To: <sip:(spam-protected);user=phone>;tag=as06405fd1
Call-ID: 5412a02ed9ab-qyx8rx3u3y5p
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:(spam-protected):2049;line=pqjgirdk>;reg-id=1
Content-Length: 0

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