[Users] Issues with calls using openser.
Shane Burrell
shaneb at metrostat.net
Wed Jan 3 22:25:01 CET 2007
I changed the codec to ULAW which is defiantly supported. I'm thinking the
reINVITE may be the problem but I'm pretty new at openser configuration and
don't see a clear way to detect a reinvite and not auth it. I did capture a
ngrep of a failed call. I'll also test xlite or a sipura tonight to see if
it something specific to the grandstream.
ngrep -q -t -W byline port 5060
interface: eth0 (192.168.16.0/255.255.255.0)
filter: (ip) and ( port 5060 )
U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060
INVITE sip:7005874200 at siprt1.siptest.net:5060;user=phone SIP/2.0.
t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
k: replaces.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 249.
.
v=0.
o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 0 101 .
a=silenceSupp:off.
a=ecan:b on g168.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 100 Giving a try.
t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14845
req_src_ip=192.168.16.91 req_src_port=5060
in_uri=sip:7005874200 at siprt1.siptest.net:5060;user=phone
out_uri=sip:7005874200 at 192.168.17.83;user=phone via_cnt==1".
.
U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060
INVITE sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
k: replaces.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 249.
.
v=0.
o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 0 101 .
a=silenceSupp:off.
a=ecan:b on g168.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.
U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.
U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Content-Length: 0.
.
U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
Content-Length: 220.
.
v=0.
o=7005874200 8000 8000 IN IP4 192.168.17.83.
s=SIP Call.
c=IN IP4 192.168.17.83.
t=0 0.
m=audio 10000 RTP/AVP 18 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 INVITE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
Content-Length: 220.
.
v=0.
o=7005874200 8000 8000 IN IP4 192.168.17.83.
s=SIP Call.
c=IN IP4 192.168.17.83.
t=0 0.
m=audio 10000 RTP/AVP 18 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060
ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 ACK.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200 at 192.168.17.83;user=phone>.
User-Agent: MSTSYLVAIPGW.
l: 0.
.
U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060
ACK sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979699 ACK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
User-Agent: MSTSYLVAIPGW.
l: 0.
.
U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060
INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
Remote-Party-Id:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off.
Proxy-Require: privacy.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979700 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200 at 192.168.17.83;user=phone>.
m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
c: application/sdp.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 236.
.
v=0.
o=MSTNT 536702936 536702937 IN IP4 192.168.16.91.
s=Session SDP.
c=IN IP4 192.168.16.91.
t=0 0.
m=audio 40878 RTP/AVP 18 101.
a=silenceSupp:off.
a=ecan:b on g168.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 407 Proxy Authentication Required.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979700 INVITE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Proxy-Authenticate: Digest realm="192.168.16.91",
nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5".
Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14844
req_src_ip=192.168.16.91 req_src_port=5060
in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598
out_uri=sip:7005874200 at 192.168.17.83;user=phone via_cnt==1".
.
U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060
ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979700 ACK.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
User-Agent: MSTSYLVAIPGW.
l: 0.
.
U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060
BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979701 BYE.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 70.
Route: <sip:7005874200 at 192.168.17.83;user=phone>.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 0.
.
U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060
BYE sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
t:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
f:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979701 BYE.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
v: SIP/2.0/UDP 192.168.16.91:5060.
Max-Forwards: 69.
Accept: application/sdp.
Accept-Encoding: .
Accept-Language: en.
User-Agent: MSTSYLVAIPGW.
l: 0.
.
U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979701 BYE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Supported: replaces.
Content-Length: 0.
.
U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.16.91:5060.
Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
From:
<sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
8.
To:
<sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
CSeq: 12979701 BYE.
User-Agent: Grandstream HT496 1.0.3.64 FXS0.
Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Supported: replaces.
Content-Length: 0.
.
U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060
..................
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
Sent: Wednesday, January 03, 2007 7:51 AM
To: Shane Burrell
Cc: users at openser.org
Subject: Re: [Users] Issues with calls using openser.
The log shows that you challenge reINVITEs. MAybe this breaks the
grandstream. Please try without challenging the reINVITE. If this helps,
then it is a probably a grandstream bug.
But of course there is the question why the grandstream sends a reINVITE
at all? This is often a codec problem.
Please also try to use other client (e.g. xlite) and post complete ngrep
dumps: "ngrep -q -t -W byline port 5060"
regards
klaus
Shane Burrell wrote:
> The SIP UA was a grandstream ATA running the latest stable firmware.
Prior
> to upgrading to 1.1 and moving to mediaproxy it worked well with the
> exception of good nat support which is why I would really like mediaproxy
to
> work. Is there anything I should look for in the sip dialog to determine
if
> the client, sip proxy, or the gateway is the culprit on disconnecting the
> call?
>
> Thanks,
>
> Shane
>
> -----Original Message-----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> Sent: Wednesday, January 03, 2007 4:52 AM
> To: Shane Burrell
> Cc: users at openser.org
> Subject: Re: [Users] Issues with calls using openser.
>
>
> Maybe a bug in the caller's SIP client?
>
> regards
> klaus
>
> Shane Burrell wrote:
>> I recently installed the latest version of openser and this time used
>> mediaproxy rather than rtpproxy. Everything seems to work but if a sip
>> device is called the phone rings and is instantally disconnected and the
> far
>> end is left off-hook. This worked before but I did modify my script to
> work
>> with mediaproxy. Below is the wireshark decode of the sip messagining.
> Any
>> help or suggestions on where to look would be great. Calls from the sip
>> device works flawlessly. I am using a MaxTNT as the gateway.
>>
>>
>>
>>
>>
>> |Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
>>
>> |22.031 | INVITE SDP ( telephone-event) |
>> |SIP From: sip:8385021101 at 192.168.16.91:5060 To:sip: 8385024200@
>> siprt1.me.net:5060
>>
>> | |(5060) ------------------> (5060) | |
>>
>> |22.031 | 100 Giving a try |
|SIP
>> Status
>>
>> | |(5060) <------------------ (5060) | |
>>
>> |22.031 | | INVITE SDP ( telephone-event)
>> |SIP Request
>>
>> | | |(5060) ------------------> (5060) |
>>
>> |22.040 | | 100 Trying|
|SIP
>> Status
>>
>> | | |(5060) <------------------ (5060) |
>>
>> |22.042 | | 180 Ringing
|SIP
>> Status
>>
>> | | |(5060) <------------------ (5060) |
>>
>> |22.042 | 180 Ringing |
|SIP
>> Status
>>
>> | |(5060) <------------------ (5060) | |
>>
>> |25.244 | | 200 OK SDP ( telephone-event)
>> |SIP Status
>>
>> | | |(5060) <------------------ (5060) |
>>
>> |25.245 | 200 OK SDP ( telephone-event) |
>> |SIP Status
>>
>> | |(5060) <------------------ (5060) | |
>>
>> |25.269 | ACK | |
|SIP
>> Request
>>
>> | |(5060) ------------------> (5060) | |
>>
>> |25.269 | | ACK |
|SIP
>> Request
>>
>> | | |(5060) ------------------> (5060) |
>>
>> |25.269 | INVITE SDP ( telephone-event) |
>> |SIP From: sip: 8385021101 at 152.93.36.91:5060 To:sip: 8385024200@
>> siprt1.me.net:5060
>>
>> | |(5060) ------------------> (5060) | |
>>
>> |25.270 | 407 Proxy Authentication Required |
>> |SIP Status
>>
>> | |(5060) <------------------ (5060) | |
>>
>> |25.291 | ACK | |
|SIP
>> Request
>>
>> | |(5060) ------------------> (5060) | |
>>
>> |25.291 | BYE | |
|SIP
>> Request
>>
>> | |(5060) ------------------> (5060) | |
>>
>> |25.293 | | BYE |
|SIP
>> Request
>>
>> | | |(5060) ------------------> (5060) |
>>
>> |25.326 | | 200 OK |
|SIP
>> Status
>>
>> | | |(5060) <------------------ (5060) |
>>
>> |25.327 | 200 OK | |
|SIP
>> Status
>>
>> | |(5060) <------------------ (5060) | |
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Shane
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Users mailing list
>> Users at openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
--
Klaus Darilion
nic.at
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