"400 BAD REQUEST" bei INVITE

[MarcelDeLuX]

Servus,

musste gerade mal ein Telefon austauschen (Grandstream GXP 2000) und dabei kam ein Problem auf, was ich nicht wirklich durchschaue...

Habe im Telefon ordnungsgemäß die Zugangsdaten eingetragen, Telefon registriert sich auch ohne Probleme.
Interne und ausgehende Telefonate gehen auch, aber eingehend bekomme ich einen Fehler, der nach dem SIP-Protokoll folgendes zu Tage bringt:

<-- SIP read from 192.168.100.218:5060:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK2af9b067;rport
From: "DOKOM<017252XXX>" <sip:[email protected]>;tag=as6c920e84
To: <sip:[email protected]:5060;transport=udp>;tag=7ce92321eeb22fae
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.2.2.26
Warning: 398 "You are not allowed to CANCEL an established dialog"
Content-Length: 0

Der gesamte SIP-Verkehr vom Anruf:

  == dokom#01: Incoming call '017252XXX' -> '91722918'
    -- dokom#01: Updated channel name: CAPI/dokom#01/91722918-34
    -- Executing SetCallerPres("CAPI/dokom#01/91722918-34", "allowed") in new stack
    -- Executing Set("CAPI/dokom#01/91722918-34", "CALLERID(name)=DOKOM<017252XXX>") in new stack
    -- Executing Dial("CAPI/dokom#01/91722918-34", "SIP/18") in new stack
We're at 192.168.100.200 port 11252
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 22 lines
Reliably Transmitting (no NAT) to 192.168.100.218:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK2af9b067;rport
From: "DOKOM<017252XXX>" <sip:[email protected]>;tag=as6c920e84
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 13 Aug 2010 11:21:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 516

v=0
o=root 7962 7962 IN IP4 192.168.100.200
s=session
c=IN IP4 192.168.100.200
t=0 0
m=audio 11252 RTP/AVP 8 0 4 3 111 5 10 7 18 110 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 18
asterisk1*CLI>
<-- SIP read from 192.168.100.218:5060:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK2af9b067;rport
From: "DOKOM<017252XXX>" <sip:[email protected]>;tag=as6c920e84
To: <sip:[email protected]:5060;transport=udp>;tag=7ce92321eeb22fae
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.2.2.26
Warning: 398 "You are not allowed to CANCEL an established dialog"
Content-Length: 0


--- (9 headers 0 lines) ---
    -- Got SIP response 400 "Bad Request" back from 192.168.100.218
Transmitting (no NAT) to 192.168.100.218:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK2af9b067;rport
From: "DOKOM<017252XXX>" <sip:[email protected]>;tag=as6c920e84
To: <sip:[email protected]:5060;transport=udp>;tag=7ce92321eeb22fae
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/18-0913af68 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Set("CAPI/dokom#01/91722918-34", "CALLERID(all)=von DOKOM<017252XXX><017252XXX>") in new stack
    -- Executing Dial("CAPI/dokom#01/91722918-34", "SIP/13|15") in new stack
...

nach "..." kommt die Umleitung bei besetzt, die dann auch normal durchläuft...

Wo könnte das Problem liegen???
BITTE HELFT MIR :doof:

 

[MarcelDeLuX]

okay, lag doch nicht an mir, hab die neuste Firmware aufs telefon gezogen und alles funktioniert wie gewollt,

Von daher: :toilet: