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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Voipstunt + OKI ATA troubles



 
 
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  #1  
Old June 27th 06, 01:16 PM posted to uk.telecom.voip
Jan De Luyck
external usenet poster
 
Posts: 3
Default Voipstunt + OKI ATA troubles

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

  #2  
Old June 28th 06, 03:03 AM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 16
Default Voipstunt + OKI ATA troubles

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

  #3  
Old June 28th 06, 07:08 PM posted to uk.telecom.voip
Jan De Luyck
external usenet poster
 
Posts: 3
Default Voipstunt + OKI ATA troubles

On 2006-06-28, Jon Farmer wrote:
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?


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 ;tag=3381863556
To: voipstunt
Call-ID:
CSeq: 14 REGISTER
Contact: :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 ;tag=3381863556
To: voipstunt
Contact: sip:194.120.0.202:5060
Call-ID:
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 ;tag=3381863556
To: voipstunt
Contact: sip:192.168.33.150:5060
Call-ID:
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 ;tag=3381863556
To: voipstunt
Contact: sip:80.239.235.201:5060
Call-ID:
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 ;tag=1003685425
To: voipstunt
Call-ID:
CSeq: 16 REGISTER
Contact: :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 ;tag=1003685425
To: voipstunt
Contact: sip:194.221.62.206:5060
Call-ID:
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 ;tag=1003685425
To: voipstunt
Contact: sip:192.168.33.150:5060
Call-ID:
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.
Send to 127.0.0.1:2727 ---

NTFY 306 ] MGCP 1.0
X: 10
O: x,x,x,x,x,x,x,x,x,x,x,x,T

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 ] 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", sff
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
Send to 127.0.0.1:2727 ---

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

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/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt ;tag=3212241778
To:
Call-ID:
CSeq: 300 INVITE
Contact: :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 ] MGCP 1.0
X: 11
R: hu


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


Parser status 200
Send to 127.0.0.1:2727 ---

200 3016 OK

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 ;tag=3212241778
To:
Contact: :5060
Call-ID:
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/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060
From: voipstunt ;tag=3212241778
To:
Call-ID:
CSeq: 300 ACK
Content-Length: 0


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


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


Parser status 200
MGCP DELETE CNX: endpt 0, cnx 0
Send to 127.0.0.1:2727 ---

250 3017 Connection was deleted
P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=0, LA=0

Recv from 127.0.0.1:2727 ---
RQNT 3018 ] 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
Send to 127.0.0.1:2727 ---

200 3018 OK

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
Send to 127.0.0.1:2727 ---

NTFY 307 ] MGCP 1.0
X: 12
O: hu

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 ] MGCP 1.0
X: 13
R: hd
S:


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


Parser status 200
Send to 127.0.0.1:2727 ---

200 3019 OK

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 ;tag=2059885872
To: voipstunt
Call-ID:
CSeq: 17 REGISTER
Contact: :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 ;tag=2059885872
To: voipstunt
Contact: sip:194.221.62.207:5060
Call-ID:
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 ;tag=2059885872
To: voipstunt
Call-ID:
CSeq: 18 REGISTER
Contact: :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 ;tag=2059885872
To: voipstunt
Contact: sip:194.120.0.202:5060
Call-ID:
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 ;tag=2059885872
To: voipstunt
Contact: sip:192.168.33.150:5060
Call-ID:
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?

  #4  
Old June 28th 06, 07:23 PM posted to uk.telecom.voip
Jon Farmer
external usenet poster
 
Posts: 16
Default Voipstunt + OKI ATA troubles

Jan De Luyck wrote:
On 2006-06-28, Jon Farmer wrote:
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?


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 ;tag=1003685425
To: voipstunt
Call-ID:
CSeq: 16 REGISTER
Contact: :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
  #5  
Old June 30th 06, 08:25 AM posted to uk.telecom.voip
Jan De Luyck
external usenet poster
 
Posts: 3
Default Voipstunt + OKI ATA troubles

On 2006-06-28, Jon Farmer wrote:
Jan De Luyck wrote:
--- 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 ;tag=1003685425
To: voipstunt
Call-ID:
CSeq: 16 REGISTER
Contact: :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?


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
 




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