SIP: 420 Bad Extension

[KarstenHuhn]

Hi guys,

system (192.168.0.91) is a MiVB7.1 with Vodafone GER SIP trunk (formerly Arcor).
The system uses no MBG and communicates directly with the gateway 192.168.0.1.
All settings were done according to SIP COE Mitel with VF GER.
The system registers and the SIP number can be called, both way audio -> OK.

But when trying to perform an outgoing call from IP Phone 192.168.0.94 no single call can be completed.
Every trial results in a "420 Bad Extension" (see TCPDUMP below - please note that **** were filled in for security reasons).

Provider cannot help, Mitel cannot help - can you?
Which setting(s) should possibly be modified?

Any help would be appreciated.

Thanks,
Karsten

TCPDUMP:
--------

¡²ÃÔ   ÿÿ
U[†Ž ~ð ¡ ¡ € ¾ï  ºº Eº“ A @qšÀ¨ [²’²ÄÄ REGISTER sip:0202.sip.arcor.de:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.91:5060;branch=z9hG4bK3522346608-103060487
Max-Forwards: 70
From: <sip:0202****202@0202.sip.arcor.de>;tag=0_3431766608-103060473
To: <sip:0202****202@0202.sip.arcor.de>
Call-ID: 3431766608-103060471
CSeq: 5 REGISTER
Authorization: Digest username="0202****202",realm="arcor.de",nonce="555b8634f6b7b1c21e0330e3fad23b3ff8d71f7a",uri="sip:0202.sip.arcor.de:5060",response="10fa0aff14be9ea7103e0e5f5a1a6b8e",algorithm=MD5
Contact: <sip:0202****202@192.168.0.91:5060;transport=udp>
Expires: 3600
User-Agent: Mitel-3300-ICP 13.1.0.33
Content-Length: 0

U[†Ž 0
«
«  ºº € ¾ï Eº
 A @r²’²À¨ [ÄÄ
‰ SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.91:5060;received=178.0.151.126;branch=z9hG4bK3522346608-103060487;rport=64491
From: <sip:0202****202@0202.sip.arcor.de>;tag=0_3431766608-103060473
To: <sip:0202****202@0202.sip.arcor.de>;tag=aprquc1l8b3-ut24fa30000a0
Call-ID: 3431766608-103060471
CSeq: 5 REGISTER
Contact: <sip:0202****202@192.168.0.91:5060;transport=udp>;expires=60

U[† Ið . .  ºº € ¾ï Eº A @»Ù
O À¨ [ÄÄ  U[† EÀ . .  ºº € ¾ï Eº A @»Ù
O À¨ [ÄÄ  U[† ~ð O O € ¾ï  ºº EºA A @oìÀ¨ [²’²ÄÄ- INVITE sip:0210220786190@0202.sip.arcor.de:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.91:5060;branch=z9hG4bK3537336608-103060489
Max-Forwards: 26
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Require: 100rel
From: "Schreibtisch EG" <sip:0202****202@192.168.0.91>;tag=0_3537336608-103060490
To: <sip:0210220786190@0202.sip.arcor.de>
Call-ID: 3537336608-103060488
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 90;Refresher=uac
Supported: replaces,timer
Contact: "Schreibtisch EG" <sip:0202****202@192.168.0.91:5060;transport=udp>
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 13.1.0.33
P-Asserted-Identity: "Schreibtisch EG" <sip:0202****202@192.168.0.91:5060;transport=udp>
Content-Length: 289

v=0
o=- 5355 5355 IN IP4 192.168.0.94
s=-
c=IN IP4 192.168.0.94
t=0 0
m=audio 50308 RTP/AVP 102 8 0 18 121 101
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:121 L16/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
U[† 0

  ºº € ¾ï Eº
 A @r¬²’²À¨ [ÄÄ
m SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 192.168.0.91:5060;received=178.0.151.126;branch=z9hG4bK3537336608-103060489;rport=64491
From: "Schreibtisch EG" <sip:0202****202@192.168.0.91>;tag=0_3537336608-103060490
To: <sip:0210220786190@0202.sip.arcor.de>;tag=aprqngfrt-lcune33000020
Call-ID: 3537336608-103060488
CSeq: 1 INVITE
Unsupported: 100rel

U[†ž
ð
ð € ¾ï  ºº Eº
â A @rKÀ¨ [²’²ÄÄ
Î ACK sip:0210220786190@0202.sip.arcor.de:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.91:5060;branch=z9hG4bK3537336608-103060489
Max-Forwards: 70
From: "Schreibtisch EG" <sip:0202****202@192.168.0.91>;tag=0_3537336608-103060490
To: <sip:0210220786190@0202.sip.arcor.de>;tag=aprqngfrt-lcune33000020
Call-ID: 3537336608-103060488
CSeq: 1 ACK
Contact: "Schreibtisch EG" <sip:0202****202@192.168.0.91:5060;transport=udp>
Content-Length: 0

 

[Billz66]

What do you mean by this
"The system registers and the SIP number can be called, both way audio -> OK."

Does this man that incoming calls are ok but outgoing are failing?

if so make sure that you are sending correct Calling Line Information


If I never did anything I'd never done before , I'd never do anything.....

and due to an endless stream of MiCollab , MiCC issues
Life would be simpler If only they tested products properly before releasing them .....

 

[KarstenHuhn]

That's the point, Billz66, I'm pretty sure to send the correct calling line information, I tried every variant.
And yes, you're right, incoming calls are OK, outgoing are failing.

 

[KarstenHuhn]

The problem is resolved:
The flag "Require Reliable Provisional Responses on Outgoing Calls" was set to YES but had to be NO.

Karsten