[GnomeMeeting-list] Security check fail 2 ;(

["W.P." <laurentp poczta onet pl>]

Configuration:
Ekiga 2.0.1 installed from rpm, Kernel 2.6.13, ALSA, PWLib & Opal 
installed from Ekiga site, Duron, 256RAM. OS Aurox (Fedora Core 3/4). 
Have created accounts on Actio (www.actio.pl), Ekiga.Net, and Halo 
(proxy.softphone.pl). Have configured Ekiga with instructions from Actio 
operator, there are funds on this account, to permit dial-out to PSTN. 
Actio is default account and also has a dial-in number in PSTN. 
Connection to Internet via GPRS modem, public IP address, no NAT. Also 
installed suggested by Actio X-Lite, not using both at the sime time, of 
course.
Ekiga conf: Personal data - empty, but I tested with my real name and 
with Actio account name, no difference. Network settings: No NAT 
transversal, listening on ppp0 (shows correct address). SIP settings: 
outbound proxy : sip.actio.pl, forward: sip: (tested also with 
sip:ActioName, and sip:ActioName sip actio pl). Codec settings: all 
enabled with GSM codec at 3-rd place. On firewall (iptables) enabled UDP 
ports 5000:5100, TCP also (that shown in H.323 config).

I'm aware, that having only GPRS not CDMA may lead to problems with 
voice quality.

Successfully registers in all accounts (sometimes timeout, but repeating 
works).

Problem(s):

1). Dial-in from other phone to Actio-provided PSTN number works 
(signalling is OK), CLIP present. I can hear in my speakers what I'm 
saying to "real" telephone mic. "Real" phone is an GSM headset (not that 
one used as modem). But it works only one-way.. -> NO any sound in 
"real" phone speaker when talking to PC. Mixer is enabled, recording is 
enabled, and traffic stats shows I'm transmitting voice. Is that Ekiga 
or operator problem? How to identify? When I DON'T set outbound proxy, I 
can connect to ex. echo proxy01 softphone com, hear the message and hear 
myself (very poor quality, but I think it's codec/bandwith problem).

2). (MORE SERIOUS). Outbound calls: When I DONT set outbound proxy, 
calls to SIP adresses are succesful. When I try to call PSTN phone 
(actually my other GSM handset i got "Security check fail". Same for 
trying to dial echo server.
      I've run trace level 4 for Ekiga, and trace for other soft 
(X-Lite). BTW, X-Lite registers, calls-out (at least signalling, but no 
voice), but cannot accept             incoming call (but this is NTG ;)

Traces show some differences at SIP signalling level. Here it goes:

----------------------- BEGIN X-Lite debug --------------------------------
>>> REGISTER with:
....
....
From: being my account number<my SIP address),
To: same,
Call-ID: some-string sip actio pl

<<< 401 Unauthorized with:
.....
.....
WWW-Authenticate: Digest realm="sip.actio.pl" nonce="some-string"
Server: Sip Express router (0.9.4 i386/freebsd)

>>>> REGISTER with:
.....
.....
username="my-Actio-username" realm="sip.actio.pl" nonce="......" 
response="......."

<<<< 200 OK with:
.....
.....
.....

>>> ACK
....
....
....

>>> INVITE with:
From: my-Actio-name<my-actio-name sip actio pl)
To: <sip:0GSMnumber sip actio pl>
Authorization: username="my-Actio-name", realm="81.15.209.199" (IP of 
sip.actio.pl) nonce="....." response="....." 
uri=sip:0GSMnumber sip actio pl"
....
....
....

<<< 100 trying
....
....
....

<<< 183 session progress
....
....
....

At this moment (I think) GSM rings, I accept the connection (but hear 
nothing), next I cancel the call.

>>> CANCEL

----------------------- END X-Lite debug --------------------------------

----------------------- BEGIN Ekiga debug --------------------------------

>>> REGISTER with:
....
....
From: being my account number<my SIP address),
To: same,
Call-ID: some-string laurent-home (my computer name) <THERE IS DIFFERENCE>

<<< 401 Unauthorized with:
.....
.....
WWW-Authenticate: Digest realm="sip.actio.pl" nonce="some-string"
Server: Sip Express router (0.9.4 i386/freebsd)

>>>> REGISTER with:
.....
.....
username="my-Actio-username" realm="sip.actio.pl" nonce="......" 
response="......."

<<<< 200 OK with:
.....
.....
.....

several SUBSCRIBE packets with response 501 Not implemented.

now I try to call-out......

GMURLHandler SIP No authentication information present < THIS MAY BE 
IMPORTANT>-----------------------------------

>>> INVITE with:
From: my-Actio-name<my-actio-name sip actio pl)
To: <sip:0GSMnumber sip actio pl>
<NO AUTHORISATION INFORMATION><THIS IS THE 
DIFFERENCE>-------------------------------------
Call-ID: some-string laurent-home (my computer name) <THERE IS DIFFERENCE>
....
....
....

<<<<  100 Trying

<<<< 401 Unauthorized
.....
.....
WWW-Authenticate: Digest realm="81.15.209.199" nonce="some-string" 
<DIFFERENCE TO packet received during REGISTER!!!!!!, there realm was in 
string>

>>>> ACK

-------------------------------------------------------------------------- 
MAY MOST IMPORTANT FRAGMENT------------------------------------
SIP Received Authentication Req Response
SIP Coundn't fint authenication information on realm 81.15.209.199, will 
use SIP outbound Proxy auth setting, if any.
OpalCon Releasing Call
OpanCon Call end reason for ..... is Ended By Security Denial.....
....
SIP Cannot do Authentication Required for non REGISTER, SUBSCRIBE or 
MESSAGE.
.....
----------------------- END Ekiga debug --------------------------------