Voice Over IP in UK - Voipstunt + OKI ATA troubles

Author

Voipstunt + OKI ATA troubles

Jan De Luyck

2006-06-27, 1:11 pm

Hello all,

I've got a subscription with voipstunt, and I've been using it for three months
without any troubles using my OKI BMG7012.

Since the 17th tho, I am unable to place any call using my ATA adapter. Using
a softphone (twinkle on linux) works perfectly, so there are no troubles in my
home setup.

I've traced the outgoing sip conversation, and as I dial out a number, I get an
401 Unauthorized back from the voipstunt and I get a busy tone from the ATA.

Any ideas what I could check? I've verified all the settings (I've re-entered
them this morning after doing a full reset), the internet connection is okay
(normally it's hooked up behind a wifi router, but I connected it straight on
the net this morning too), so I'm a bit stumped what to do next.

Voipstunt themselves "don't officially support SIP", and they don't respond to
mails - atleast sofar not.

Thanks,

Jan

--
Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
"To IBM, 'open' means there is a modicum of interoperability among some of their
equipment."
-- Harv Masterson

Jon Farmer

2006-06-28, 1:11 am

Jan De Luyck wrote:
> Hello all,
>
> I've got a subscription with voipstunt, and I've been using it for three months
> without any troubles using my OKI BMG7012.
>
> Since the 17th tho, I am unable to place any call using my ATA adapter. Using
> a softphone (twinkle on linux) works perfectly, so there are no troubles in my
> home setup.
>
> I've traced the outgoing sip conversation, and as I dial out a number, I get an
> 401 Unauthorized back from the voipstunt and I get a busy tone from the ATA.

Can you post the SIP messages here so we can see what is going on?

> Voipstunt themselves "don't officially support SIP", and they don't respond to
> mails - atleast sofar not.

Never used Voipstunt myself but if what you say above is true it turns
me right off them.

Regards

Jon

Jan De Luyck

2006-06-28, 7:11 pm

On 2006-06-28, Jon Farmer wrote:
> Jan De Luyck wrote:
>
> Can you post the SIP messages here so we can see what is going on?

Sure. Big post coming on, it's the actual attempt to dial a PSTN line. (my own)

I've obscured my own IP, my SIP username (username) and the number I tried to
dial.
They're all valid.

192.168.33.150 is the internal ip address of the ATA.

-------------------------------------------
---> Send to sip.voipstunt.com:5060 at tick 36798
REGISTER sip:voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3381863556
To: voipstunt <sip:username@voipstunt.com>
Call-ID: 1973839901@192.168.33.150
CSeq: 14 REGISTER
Contact: <sip:username@XX.XX.XX.XX:5060>
Authorization: Digest username="username", realm="sipdiscount.com",
nonce="878586346", uri="sip:sip.voipstunt.com:5060", response="password",
algorithm=MD5
max-forwards: 70
expires: 60
Content-Length: 0

<--- Recv from 194.120.0.202:5060 at tick 36808
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3381863556
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:194.120.0.202:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 14 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

<--- Recv from 194.120.0.202:5060 at tick 36810
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.33.150:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3381863556
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:192.168.33.150:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 14 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

VOIP led status = on.
<--- Recv from 80.239.235.201:5060 at tick 37273
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3381863556
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:80.239.235.201:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 13 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
WWW-Authenticate: Digest
realm="sipdiscount.com" ,nonce="603052190" ,algorithm=MD5
Content-Length: 0

---> Send to sip.voipstunt.com:5060 at tick 42608
REGISTER sip:voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=1003685425
To: voipstunt <sip:username@voipstunt.com>
Call-ID: 1973839901@192.168.33.150
CSeq: 16 REGISTER
Contact: <sip:username@XX.XX.XX.XX:5060>
Authorization: Digest username="username", realm="sipdiscount.com",
nonce="4003847063", uri="sip:sip.voipstunt.com:5060", response="password",
algorithm=MD5
max-forwards: 70
expires: 60
Content-Length: 0

string=0

DTMF EVENT line 0: 0
MGCP SIGNAL: endpt 0, cnx -1, evt 11 (DIALTONE) = OFF
[VoIP -> PSTN] VoIP DTMF digit Input 0.
processCmdQ: EPTCMD_SIGNAL
stopTone: devid 0 keeping tone vhd for tone det
string=0

DTMF EVENT line 0: 0
[VoIP -> PSTN] VoIP DTMF digit Input 0.
<--- Recv from 194.221.62.206:5060 at tick 42697
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=1003685425
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:194.221.62.206:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 16 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

<--- Recv from 194.221.62.206:5060 at tick 42700
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.33.150:5060
From: voipstunt <sip:username@voipstunt.com>;tag=1003685425
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:192.168.33.150:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 16 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

VOIP led status = on.
string=5

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=3

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=3

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=3

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=3

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=3

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=4

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=1

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=0

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=0

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=8

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
string=6

DTMF EVENT line 0: x
[VoIP -> PSTN] VoIP DTMF digit Input x.
[VoIP -> PSTN] VoIP DTMF digit Input T.
[VoIP -> PSTN] VoIP digit string dial out.[vbcol=seagreen]
NTFY 306 aaln/1@[192.168.33.150] MGCP 1.0
X: 10
O: x,x,x,x,x,x,x,x,x,x,x,x,T[vbcol=seagreen
]
<<< Recv from 127.0.0.1:2727 ---
200 306 OK
<<<

A: 9 edge down.
======== Alan debug: CA has Response (1)

Parser status 200
Tmr adj to=0, RTT=58, DEV=65, RTO=188
<<< Recv from 127.0.0.1:2727 ---
CRCX 3015 AALN/1@[192.168.33.150] MGCP 1.0
C: 0
L: p:20,
a:G729A;PCMU;PCMA;G.723.1-5.3;G723.1;G723.1A-5.3;G723.1A;G729B;telephone-event,
fmtp:"telephone-event 0-15", s:off
M: recvonly
<<<

======== Alan debug: CA has Response (1)

Parser status 200
MGCP CREATE CNX: endpt 0, cnx 0
ccConnection remote (decode) codec list:
(9): 18=7:0 0=0:0 8=1:0 106=9:0
ccConnection local (encode) codec list:
(9): 18=7:0 0=0:0 8=1:0 106=9:0
ccConnection: codec: 7, period: 20, vad: 0, relay: 2
Connection Mode: RecvOnly
ccConnection: Attached device 0 to stream 0 for cxid 0[vbcol=seagreen]
200 3015 OK
I: 5A51

v=0
o=Broadcom 23121 3015 IN IP4 XX.XX.XX.XX
s=MGCP call
c=IN IP4 XX.XX.XX.XX
b=AS:64
t=0 0
m=audio 5004 RTP/AVP 18 0 8 106 4 107 108 109 110
a=rtpmap:106 G.723.1-5.3/8000
a=rtpmap:107 G723.1A-5.3/8000
a=rtpmap:108 G723.1A/8000
a=rtpmap:109 G729B/8000
a=rtpmap:110 TELEPHONE-EVENT/8000
a=fmtp:110 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15
a=recvonly
a=ptime:20[vbcol=seagreen]
processCmdQ: EPTCMD_CREATE_STREAM
processCmdQ: EPTCMD_ATTACH_DEVICE
streamDevAttach: netvhd NULL for stream 0
streamDevAttach: convert tone vhd to net vhd for dev 0
HDSP: endpt 80 mode change event op1:0 op2:0

HDSP: endpt 80 mode change event op1:2 op2:0

length: application/sdp 15 18
---> Send to sip.voipstunt.com:5060 at tick 43256
INVITE sip:xxxxxxxxxxxx@voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3212241778
To: <sip:xxxxxxxxxxxx@voipstunt.com>
Call-ID: 3868338711@192.168.33.150
CSeq: 300 INVITE
Contact: <sip:username@XX.XX.XX.XX:5060>
max-forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 167

v=0
o=- 0 0 IN IP4 XX.XX.XX.XX
s=-
c=IN IP4 XX.XX.XX.XX
b=AS:64
t=0 0
m=audio 5004 RTP/AVP 18 0 8 4 110
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=ptime:20

number of message sent: 6
<<< Recv from 127.0.0.1:2727 ---
RQNT 3016 AALN/1@[192.168.33.150] MGCP 1.0
X: 11
R: hu
<<<

======== Alan debug: CA has Response (1)

Parser status 200[vbcol=seagreen]
200 3016 OK[vbcol=seagreen]
hdspctrl_setEcanCtrl: Setting PxD handle of VHD0 to PXD0
HDSP: endpt 80 mode change event op1:0 op2:0

HDSP: endpt 80 mode change event op1:3 op2:0, stream 0

HDSP: PVE Codec for VHD 80 encoder rate change to G.729 (Annex A) 8 kbps no VAD
2 Frames
addAttDev: store devId 0 to streamId 0 list
<--- Recv from 80.239.235.200:5060 at tick 43269
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3212241778
To: <sip:xxxxxxxxxxxx@voipstunt.com>
Contact: sip:xxxxxxxxxxxx@80.239.235.200:5060
Call-ID: 3868338711@192.168.33.150
CSeq: 300 INVITE
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
WWW-Authenticate: Digest
realm="sipdiscount.com" ,nonce="360197846" ,algorithm=MD5
Content-Length: 0

---> Send to sip.voipstunt.com:5060 at tick 43272
ACK sip:xxxxxxxxxxxx@voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=3212241778
To: <sip:xxxxxxxxxxxx@voipstunt.com>
Call-ID: 3868338711@192.168.33.150
CSeq: 300 ACK
Content-Length: 0

456xx: Call emCallDrop with EndptID[0].
<<< Recv from 127.0.0.1:2727 ---
DLCX 3017 AALN/1@[192.168.33.150] MGCP 1.0
C: 0
I: 5A51
<<<

======== Alan debug: CA has Response (1)

Parser status 200
MGCP DELETE CNX: endpt 0, cnx 0[vbcol=seagreen]
250 3017 Connection was deleted
P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=0, LA=0[vbcol=seagreen]
<<< Recv from 127.0.0.1:2727 ---
RQNT 3018 AALN/1@[192.168.33.150] MGCP 1.0
X: 12
R: hu, oc
S: bz
<<<

======== Alan debug: CA has Response (1)

Parser status 200
MGCP SIGNAL: endpt 0, cnx -1, evt 9 (BUSYTONE) = ON[vbcol=seagreen]
200 3018 OK[vbcol=seagreen]
processCmdQ: EPTCMD_DETACH_DEVICE
getAttDevNum: streamId 0 cnt = 0
streamDevDetach: convert netvhd to tonevhd
HDSP: endpt 80 mode change event op1:0 op2:0

hdspEvntCb: HAPINET_INGRESSPACKET: unknown vhdHdl 80
HDSP: endpt 80 mode change event op1:2 op2:0

processCmdQ: EPTCMD_DELETE_STREAM
streamDelete: no connection exists for streamId 0
processCmdQ: EPTCMD_SIGNAL
hdspctrl_getToneVhdp: tone vhd already connected devId 0
HDSP: Custom tone BUSY

CAS EVENT line 0: ONHOOK
[VoIP -> PSTN] VoIP on-hook signal coming.
MGCP SIGNAL: endpt 0, cnx -1, evt 9 (BUSYTONE) = OFF[vbcol=seagreen]
NTFY 307 aaln/1@[192.168.33.150] MGCP 1.0
X: 12
O: hu[vbcol=seagreen]
<<< Recv from 127.0.0.1:2727 ---
200 307 OK
<<<

======== Alan debug: CA has Response (1)

Parser status 200
Tmr adj to=0, RTT=50, DEV=63, RTO=176
<<< Recv from 127.0.0.1:2727 ---
RQNT 3019 AALN/1@[192.168.33.150] MGCP 1.0
X: 13
R: hd
S:
<<<

======== Alan debug: CA has Response (1)

Parser status 200[vbcol=seagreen]
200 3019 OK[vbcol=seagreen]
processCmdQ: EPTCMD_SIGNAL
stopTone: devid 0 keeping tone vhd for tone det
---> Send to sip.voipstunt.com:5060 at tick 48600
REGISTER sip:voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=2059885872
To: voipstunt <sip:username@voipstunt.com>
Call-ID: 1973839901@192.168.33.150
CSeq: 17 REGISTER
Contact: <sip:username@XX.XX.XX.XX:5060>
max-forwards: 70
expires: 60
Content-Length: 0

<--- Recv from 194.221.62.207:5060 at tick 48605
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=2059885872
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:194.221.62.207:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 17 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
WWW-Authenticate: Digest
realm="sipdiscount.com" ,nonce="878704455" ,algorithm=MD5
Content-Length: 0

---> Send to sip.voipstunt.com:5060 at tick 48609
REGISTER sip:voipstunt.com SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=2059885872
To: voipstunt <sip:username@voipstunt.com>
Call-ID: 1973839901@192.168.33.150
CSeq: 18 REGISTER
Contact: <sip:username@XX.XX.XX.XX:5060>
Authorization: Digest username="username", realm="sipdiscount.com",
nonce="878704455", uri="sip:sip.voipstunt.com:5060", response="password",
algorithm=MD5
max-forwards: 70
expires: 60
Content-Length: 0

<--- Recv from 194.120.0.202:5060 at tick 48621
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt <sip:username@voipstunt.com>;tag=2059885872
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:194.120.0.202:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 18 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

<--- Recv from 194.120.0.202:5060 at tick 48622
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.33.150:5060
From: voipstunt <sip:username@voipstunt.com>;tag=2059885872
To: voipstunt <sip:username@voipstunt.com>
Contact: sip:192.168.33.150:5060
Call-ID: 1973839901@192.168.33.150
CSeq: 18 REGISTER
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Length: 0

VOIP led status = on.
----------------------------------------

Thanks for any insights you people might give me.

Jan
--
Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
Q: How was Thomas J. Watson buried?

Jon Farmer

2006-06-29, 1:11 am

Jan De Luyck wrote:
> On 2006-06-28, Jon Farmer wrote:
>
> Sure. Big post coming on, it's the actual attempt to dial a PSTN line. (my own)
>
> I've obscured my own IP, my SIP username (username) and the number I tried to
> dial.
> They're all valid.

> ---> Send to sip.voipstunt.com:5060 at tick 42608
> REGISTER sip:voipstunt.com SIP/2.0
> Via: SIP/2.0/UDP XX.XX.XX.XX:5060
> From: voipstunt <sip:username@voipstunt.com>;tag=1003685425
> To: voipstunt <sip:username@voipstunt.com>
> Call-ID: 1973839901@192.168.33.150
> CSeq: 16 REGISTER
> Contact: <sip:username@XX.XX.XX.XX:5060>
> Authorization: Digest username="username", realm="sipdiscount.com",
> nonce="4003847063", uri="sip:sip.voipstunt.com:5060", response="password",
> algorithm=MD5
> max-forwards: 70
> expires: 60
> Content-Length: 0

Well it looks like the device is not sending back the REGISTER message
that the server is expecting. It is a MD5 response which means both the
nonce and the response should be MD5 hashes. In the messages you post
all the nonce and response are non-MD5.

Are you sure the device you are trying to use support WWW-Authentication?

HTH

Regards

Jon

Jan De Luyck

2006-06-30, 7:11 am

On 2006-06-28, Jon Farmer wrote:
> Jan De Luyck wrote:
>
>
> Well it looks like the device is not sending back the REGISTER message
> that the server is expecting. It is a MD5 response which means both the
> nonce and the response should be MD5 hashes. In the messages you post
> all the nonce and response are non-MD5.
>
> Are you sure the device you are trying to use support WWW-Authentication?

The password is an MD5 string, the nonce I didn't change.

I can't say I can find anything in the manual I've got about it.. nothing
about the authentication whatsoever.

Fun thing is, it _used_ to work. It just isn't working anymore.

I verified, the OKI still works perfectly with eg voipfone.

Regards,

Jan

--
Linux rutabaga 2.6.17 #1 PREEMPT Mon Jun 19 08:02:42 CEST 2006 i686 GNU/Linux
A roda adora.
-- palíndromo