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Adrian Marsh

2007-Sep-10 17:28 UTC

[asterisk-users] Siemans SIP/PSTN phone S450

Hi All,

Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see "Got SIP response 405 "Method Not Allowed" back from
192.168.3.64" but the phone seems to work ok.

Any ideas where it falls over in the SIP protocol?  I've included this
in the debug below.



ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
    -- Got SIP response 489 "Bad event" back from 192.168.3.10
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.3.64:5060:
OPTIONS sip:6627 at 192.168.3.64:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as35c7a074
To: <sip:6627 at 192.168.3.64:5060>
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 42771eef7db8c7403af5def871cb477c at 192.168.3.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Sep 2007 17:23:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport=5060
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as35c7a074
To: <sip:6627 at 192.168.3.64:5060>;tag=1624959632
Call-ID: 42771eef7db8c7403af5def871cb477c at 192.168.3.6
CSeq: 102 OPTIONS
Contact: "Adrian Marsh" <sip:6627 at 192.168.3.64:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Accept: application/sdp,application/dtmf-relay
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call '42771eef7db8c7403af5def871cb477c at 192.168.3.6'
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
REGISTER sip:some.server.com SIP/2.0
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;rport
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>
Call-ID: 3870123265 at 192_168_3_64
CSeq: 291 REGISTER
Contact: "Adrian Marsh" <sip:6627 at 192.168.3.64:5060>
Max-Forwards: 70
User-Agent: S450 IP020700000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.3.64 : 5060 (NAT)
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive
d=192.168.3.64;rport=5060
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>
Call-ID: 3870123265 at 192_168_3_64
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6627 at 192.168.3.6>
Content-Length: 0


---
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bK51fe39cf13e93bc714bfe8ea31b6b958;receive
d=192.168.3.64;rport=5060
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>;tag=as5908b79f
Call-ID: 3870123265 at 192_168_3_64
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="3960830f"
Content-Length: 0


---
Scheduling destruction of call '3870123265 at 192_168_3_64' in 15000 ms
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
REGISTER sip:some.server.com SIP/2.0
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;rport
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>
Call-ID: 3870123265 at 192_168_3_64
CSeq: 292 REGISTER
Contact: "Adrian Marsh" <sip:6627 at 192.168.3.64:5060>
Authorization: Digest username="6627", realm="asterisk",
algorithm=MD5,
uri="sip:some.server.com", nonce="3960830f",
response="7e032e9766f943e9f60f7d1f46114dee"
Max-Forwards: 70
User-Agent: S450 IP020700000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.3.64 : 5060 (NAT)
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received
=192.168.3.64;rport=5060
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>
Call-ID: 3870123265 at 192_168_3_64
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6627 at 192.168.3.6>
Content-Length: 0


---
Transmitting (NAT) to 192.168.3.64:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.64:5060;branch=z9hG4bKca6e25fd9fe65366c967bc15f17a7b1;received
=192.168.3.64;rport=5060
From: Adrian Marsh <sip:6627 at some.server.com>;tag=3054246604
To: Adrian Marsh <sip:6627 at some.server.com>;tag=as5908b79f
Call-ID: 3870123265 at 192_168_3_64
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 180
Contact: <sip:6627 at 192.168.3.64:5060>;expires=180
Date: Mon, 10 Sep 2007 17:23:10 GMT
Content-Length: 0


---
Scheduling destruction of call '3870123265 at 192_168_3_64' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (NAT) to 192.168.3.64:5060:
NOTIFY sip:6627 at 192.168.3.64:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK2fa265b3;rport
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as539ed18c
To: <sip:6627 at 192.168.3.64:5060>
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 0460e7f3485739803580767241924e79 at 192.168.3.6
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:voicemail at 192.168.3.6
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of call
'0460e7f3485739803580767241924e79 at 192.168.3.6' in 15000 ms
Retransmitting #1 (NAT) to 192.168.3.64:5060:
NOTIFY sip:6627 at 192.168.3.64:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK2fa265b3;rport
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as539ed18c
To: <sip:6627 at 192.168.3.64:5060>
Contact: <sip:asterisk at 192.168.3.6>
Call-ID: 0460e7f3485739803580767241924e79 at 192.168.3.6
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:voicemail at 192.168.3.6
Voice-Message: 0/0 (0/0)

---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK2fa265b3;rport=5060
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as539ed18c
To: <sip:6627 at 192.168.3.64:5060>
Call-ID: 0460e7f3485739803580767241924e79 at 192.168.3.6
CSeq: 102 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (8 headers 0 lines) ---
    -- Got SIP response 405 "Method Not Allowed" back from
192.168.3.64
Destroying call '0460e7f3485739803580767241924e79 at 192.168.3.6'
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK2fa265b3;rport=5060
From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as539ed18c
To: <sip:6627 at 192.168.3.64:5060>
Call-ID: 0460e7f3485739803580767241924e79 at 192.168.3.6
CSeq: 102 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '3870123265 at 192_168_3_64'
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:

--- (0 headers 0 lines) Nat keepalive ---
ubiphone*CLI>
[root at ubiphone log]#

Gordon Henderson

2007-Sep-10 19:21 UTC

head link

[asterisk-users] Siemans SIP/PSTN phone S450

On Mon, 10 Sep 2007, Adrian Marsh wrote:
> Hi All,
>
> Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
> and I see "Got SIP response 405 "Method Not Allowed" back
from
> 192.168.3.64" but the phone seems to work ok.
>
> Any ideas where it falls over in the SIP protocol?  I've included this
> in the debug below.
I have several Siemens C460IP's on various servers... They all do the same 
thing too. Doesn't seem to have any adverse effect though.

Gordon

Paul Hayes

2007-Sep-11 08:49 UTC

head link

[asterisk-users] Siemans SIP/PSTN phone S450

Adrian Marsh wrote:> Hi All,
> 
> Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
> and I see "Got SIP response 405 "Method Not Allowed" back
from
> 192.168.3.64" but the phone seems to work ok.
> 
> Any ideas where it falls over in the SIP protocol?  I've included this
> in the debug below.
> 
> 
> 
It is in response to Notify packets because the Siemens phone doesn't 
support presence at the moment.

It wont effect the operation of the phone or Asterisk at all.

cheers,
Paul.

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