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Home » Bugs » Record-Route header handling


11.03.2010 22:07:58

oldDude
oldDude
Posts: 16 		It looks like the Proxy is mangling multiple Roecord-Route entries when they appear in the following format: Record-Route: <a>, <b>

Trace below. Incoming INVITE looks ok. Trying returned from Polycom phone gets mangled.

SIP req. to : 192.168.0.121:5060 (UDP) - 3/11/2010 8:10:41 PM
INVITE sip:8101@192.168.0.121 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-6124j504P59005de
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK-P65192040j55d0e41;received=192.168.0.2
Via: SIP/2.0/UDP 192.168.0.2:5160;received=192.168.0.2;rport=5160;branch=z9hG4bKPj65551d902e0441038d0eb48b04ddda6c
Max-Forwards: 68
From: sip:234@SIPLinkUser2_1;tag=a40295d36f2f4dc1bd24402caabd6b11
To: sip:8101@192.168.0.2:5060
Contact: <sip:234@192.168.0.2:5160;transport=UDP>
Call-ID: 7327315629a1416e86319d698ceefcfc
CSeq: 8792 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, MESSAGE
Supported: replaces, 100rel, norefersub
Record-Route: <sip:192.168.0.2;lr>
Record-Route: <sip:192.168.0.1;lr>
User-Agent: SIPLink v2.1
Content-Type: application/sdp
Content-Length: 345
v=0
o=- 3477327040 3477327040 IN IP4 192.168.0.2
s=pjmedia
c=IN IP4 192.168.0.2
t=0 0
a=X-nat:0
m=audio 5160 RTP/AVP 18 4 3 0 8 101
a=rtcp:1805 IN IP4 192.168.0.1
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


SIP req. from : 192.168.0.121:5060 (UDP) - 3/11/2010 8:10:41 PM
SIP/2.0 100 Trying
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-6124j504P59005de
[Via] = SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK-P65192040j55d0e41;received=192.168.0.2
[Via] = SIP/2.0/UDP 192.168.0.2:5160;received=192.168.0.2;rport=5160;branch=z9hG4bKPj65551d902e0441038d0eb48b04ddda6c
[From] = <sip:234@SIPLinkUser2_1>;tag=a40295d36f2f4dc1bd24402caabd6b11
[To] = <sip:8101@192.168.0.2:5060>;tag=7DD30301-D5B1414A
[CSeq] = 8792 INVITE
[Call-ID] = 7327315629a1416e86319d698ceefcfc
[Contact] = <sip:8101@192.168.0.121>
[Record-Route] = <sip:192.168.0.2;lr>, <sip:192.168.0.1;lr>
[User-Agent] = PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
[Content-Length] = 0
SIP req. to : 192.168.0.2:5060 (UDP) - 3/11/2010 8:10:41 PM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK-P65192040j55d0e41;received=192.168.0.2
Via: SIP/2.0/UDP 192.168.0.2:5160;received=192.168.0.2;rport=5160;branch=z9hG4bKPj65551d902e0441038d0eb48b04ddda6c
From: <sip:234@SIPLinkUser2_1>;tag=a40295d36f2f4dc1bd24402caabd6b11
To: <sip:8101@192.168.0.2:5060>;tag=7DD30301-D5B1414A
CSeq: 8792 INVITE
Call-ID: 7327315629a1416e86319d698ceefcfc
Contact: <sip:8101@192.168.0.121>
Max-Forwards: 70
<sip:192.168.0.2;lr>
Record-Route: <sip:192.168.0.1;lr>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
Content-Length: 0

Granted it is strange that the Polycom phone formats the Record-Route Header in such a way, but I believe it is acceptable according to RFC. Perhaps TekSIP is only prepared to handle 1 Record-Route entry/line?
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12.03.2010 13:24:50

admin
admin
Administrator
Posts: 106 		Hi,

Can you test http://www.teksip.com/release/TekSIP-H.zip ? Let me know the result.

Best regards,

Yasin KAPLAN
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17.03.2010 13:34:23

oldDude
oldDude
Posts: 16 		Sorry for the late reply. Should I just be able to drop the new application into the folder where I installed the original version? I tried that, and when I tried to start the service, I got the following:

3/17/2010 12:31:24 PM - TekSIP Service is started on : 192.168.0.1:5060
3/17/2010 12:31:24 PM - 'Attributes' table is missing in TekSIP.mdb, check TekSIP.mdb.
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17.03.2010 17:44:15

admin
admin
Administrator
Posts: 106 		Is it possible you to download and install latest build from

http://www.teksip.com/release/teksip.zip ?

There is a new table called "Attributes" in TekSIP.mdb. Remove old TekSIP.mdb when installing new built.
• permalink • reply with quote
18.03.2010 20:20:23

oldDude
oldDude
Posts: 16 		Thanks, much better as you can see below: (although it is still valid, I would have trimmed off the extra space in the second Record-Route)
SIP req. from : 192.168.0.121:5060 (UDP) - 3/18/2010 6:47:21 PM
SIP/2.0 100 Trying
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-963306aad69d7717d
[Via] = SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bK96693d370761a7addd81-c316911e-f8db1b69
[Via] = SIP/2.0/UDP 192.168.0.2:5160;rport=5160;branch=z9hG4bKPj0b2d205a297d47e5a71d744e4611b73c
[From] = <sip:234@SIPLinkUser2_1>;tag=94592fa6f377409988a63d242e9d5ff6
[To] = <sip:8101@192.168.0.1:5060>;tag=3A11CAA1-839C8424
[CSeq] = 31257 INVITE
[Call-ID] = 2634b24130ef455bb945fa975d5be2a1
[Contact] = <sip:8101@192.168.0.121>
[Record-Route] = <sip:192.168.0.2:5060;lr>, <sip:192.168.0.1;lr>
[User-Agent] = PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
[Content-Length] = 0
SIP req. to : 192.168.0.2:5060 (UDP) - 3/18/2010 6:47:21 PM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bK96693d370761a7addd81-c316911e-f8db1b69
Via: SIP/2.0/UDP 192.168.0.2:5160;rport=5160;branch=z9hG4bKPj0b2d205a297d47e5a71d744e4611b73c
From: <sip:234@SIPLinkUser2_1>;tag=94592fa6f377409988a63d242e9d5ff6
To: <sip:8101@192.168.0.1:5060>;tag=3A11CAA1-839C8424
CSeq: 31257 INVITE
Call-ID: 2634b24130ef455bb945fa975d5be2a1
Contact: <sip:8101@192.168.0.121>
Max-Forwards: 70
Record-Route: <sip:192.168.0.2:5060;lr>
Record-Route: <sip:192.168.0.1;lr>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
Content-Length: 0
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18.03.2010 21:42:41

oldDude
oldDude
Posts: 16 		Looks like another problem this time in an OK message. It appears that the Proxy is adding an extra line when it breaks the Single line form into separate lines.

SIP/2.0 200 OK
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-5b61906ad0aa2eccd
[Via] = SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bK50ba6a129e0c6caddd81-f7f325a8-fd87482e
[Via] = SIP/2.0/UDP 192.168.0.2:5060;rport=5160;branch=z9hG4bKPj4d75f4727e704802bb108003455bacc5
[From] = <sip:234@192.168.0.2;user=phone>;tag=df4c517221a74a5fb60f3f2665318043
[To] = "8101" <sip:8101@192.168.0.2>;tag=8351B6F8-E45D22A3
[CSeq] = 17677 BYE
[Call-ID] = 27e516d4-51ff2856-a0ce68d1@192.168.0.121
[Contact] = <sip:8101@192.168.0.121>
[Record-Route] = <sip:192.168.0.2:5060;lr>, <sip:192.168.0.1;lr>
[User-Agent] = PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
[Content-Length] = 0
SIP req. to : 192.168.0.2:5060 (UDP) - 3/18/2010 8:34:42 PM
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bK50ba6a129e0c6caddd81-f7f325a8-fd87482e
Via: SIP/2.0/UDP 192.168.0.2:5060;rport=5160;branch=z9hG4bKPj4d75f4727e704802bb108003455bacc5
From: <sip:234@192.168.0.2;user=phone>;tag=df4c517221a74a5fb60f3f2665318043
To: "8101" <sip:8101@192.168.0.2>;tag=8351B6F8-E45D22A3
CSeq: 17677 BYE
Call-ID: 27e516d4-51ff2856-a0ce68d1@192.168.0.121
Contact: <sip:8101@192.168.0.121>
Max-Forwards: 70
Record-Route: <sip:192.168.0.2:5060;lr>
Record-Route: <sip:192.168.0.1;lr>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
Content-Length: 0
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19.03.2010 09:53:54

admin
admin
Administrator
Posts: 106 		Hi,
Where does TekSIP put an extra line exactly?
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19.03.2010 13:55:23

oldDude
oldDude
Posts: 16 		Looks like it got lost when I pasted in the trace. The extra line appears after the last Record-Route header - between Record-Route and User-Agent in my case. When I see it in Wireshark, it treats everything after the blank-line as MessageBody. Causes problems as you can imagine.

Max-Forwards: 70
Record-Route: <sip:192.168.0.2:5060;lr>
Record-Route: <sip:192.168.0.1;lr>
BLANK-LINE-HERE
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
Content-Length: 0
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19.03.2010 16:11:31

admin
admin
Administrator
Posts: 106 		OK. Can you test latest built which I've posted today?
http://www.teksip.com/release/TekSIP.zip
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20.03.2010 21:52:44

oldDude
oldDude
Posts: 16 		Sure. I'll get back to you by Monday with results. Thanks for quick response!
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20.03.2010 23:04:14

admin
admin
Administrator
Posts: 106 		OK, I'll be waiting for your reply.
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21.03.2010 20:08:51

oldDude
oldDude
Posts: 16 		Still seeing the same problem. Trying Ringing,OK coming from Polycom phone to Proxy all have Record-Route header that looks like this:
SIP req. from : 192.168.0.121:5060 (UDP) - 3/21/2010 6:28:48 PM
SIP/2.0 100 Trying
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-b385240ad4945720d
[Via] = SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bKb4398455274200addff0-bc25e049-4d6a546
[Via] = SIP/2.0/UDP 192.168.0.2:5160;rport=5160;branch=z9hG4bKPj56dcaf6f7aa249118ab205f0c7c98209
[From] = <sip:234@SIPLinkUser2_1>;tag=b251df8b610143f48a43f7b2fbd5fe5c
[To] = <sip:8101@192.168.0.1:5060>;tag=518FC55-6089C4B6
[CSeq] = 21584 INVITE
[Call-ID] = 0bf63bb7afb64a7fbe76ad58d6065cd3
[Contact] = <sip:8101@192.168.0.121>
[Record-Route] = <sip:192.168.0.2:5060;lr>, <sip:192.168.0.1;lr>
[User-Agent] = PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
[Content-Length] = 0

When the proxy sends it out, I see this:

SIP req. to : 192.168.0.2:5060 (UDP) - 3/21/2010 6:28:48 PM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;received=192.168.0.2;rport=5060;branch=z9hG4bKb4398455274200addff0-bc25e049-4d6a546
Via: SIP/2.0/UDP 192.168.0.2:5160;rport=5160;branch=z9hG4bKPj56dcaf6f7aa249118ab205f0c7c98209
From: <sip:234@SIPLinkUser2_1>;tag=b251df8b610143f48a43f7b2fbd5fe5c
To: <sip:8101@192.168.0.1:5060>;tag=518FC55-6089C4B6
CSeq: 21584 INVITE
Call-ID: 0bf63bb7afb64a7fbe76ad58d6065cd3
Contact: <sip:8101@192.168.0.121>
Max-Forwards: 70
Record-Route: <sip:192.168.0.2:5060;lr>
Record-Route: <sip:192.168.0.1;lr>
CRLF - Blank Line here
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036
Content-Length: 0

It appears that the proxy is successfully breaking the single line form up into multiple lines, but is adding an extra CRLF resulting in the blank line. THis gets interpreted at the far end as end of SIP message/beginning of Mesage Body. Perhaps it would be better to just leave it on the single line form.
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23.03.2010 18:02:24

admin
admin
Administrator
Posts: 106 		I think I've gound the cause of the problem. Can you test latest built which I've posted a couple of minutes ago?

http://www.teksip.com/release/TekSIP.zip
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24.03.2010 15:54:32

oldDude
oldDude
Posts: 16 		Ok. I checked the latest build and it appears that you have solved the problem! I'm assuming that if the solution works for 2 Record-route headers, it will work for more...
Thanks again
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24.03.2010 16:53:44

admin
admin
Administrator
Posts: 106 		You welcome, yes it'll work also for more than two records.
• permalink • reply with quote
11.05.2010 11:25:44

oldDude
oldDude
Posts: 16 		Me again. Is it my imagination, or is the Proxy re-ordering Record-Route headers in the following:
SIP req. from : 192.168.0.1:5061 (UDP) - 5/11/2010 9:41:48 AM
INVITE sip:123@192.168.0.1 SIP/2.0
[Via] = SIP/2.0/UDP 192.168.0.1:5061;rport;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
[From] = shawn <sip:999@192.168.0.1:5061>;tag=3543800477
[To] = <sip:123@192.168.0.1>
[Contact] = <sip:999@192.168.0.1:5061>
[Call-ID] = E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
[CSeq] = 28005 INVITE
[Max-Forwards] = 70
[Content-Type] = application/sdp
[User-Agent] = X-Lite release 1103a
[Content-Length] = 235
v=0
o=999 5117015 5117031 IN IP4 192.168.0.1
s=X-Lite
c=IN IP4 192.168.0.1
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP req. to : 192.168.0.2:5060 (UDP) - 5/11/2010 9:41:48 AM
INVITE sip:123@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7CAF9B4190022B58D
Via: SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
From: shawn <sip:999@192.168.0.1:5061>;tag=3543800477
To: <sip:123@192.168.0.2>
Contact: <sip:999@192.168.0.1:5061>
Call-ID: E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
CSeq: 28005 INVITE
Max-Forwards: 69
Record-Route: <sip:192.168.0.1;lr>
User-Agent: X-Lite release 1103a
Content-Type: application/sdp
Content-Length: 235
v=0
o=999 5117015 5117031 IN IP4 192.168.0.1
s=X-Lite
c=IN IP4 192.168.0.1
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP req. from : 192.168.0.2:5060 (UDP) - 5/11/2010 9:41:48 AM
SIP/2.0 100 Trying
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7CAF9B4190022B58D;received=192.168.0.1
[Via] = SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
[From] = shawn <sip:999@192.168.0.1:5061>;tag=3543800477
[To] = <sip:123@192.168.0.2>;tag=3700807989
[Contact] = <sip:123@192.168.0.2:5061>
[Call-ID] = E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
[CSeq] = 28005 INVITE
[Server] = X-Lite release 1103a
[Max-Forwards] = 70
[Record-Route] = <sip:192.168.0.2;lr>
[Record-Route] = <sip:192.168.0.1;lr>
[Content-Length] = 0
SIP req. to : 192.168.0.1:5061 (UDP) - 5/11/2010 9:41:48 AM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
From: shawn <sip:999@192.168.0.1:5061>;tag=3543800477
To: <sip:123@192.168.0.2>;tag=3700807989
Contact: <sip:123@192.168.0.2:5061>
Call-ID: E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
CSeq: 28005 INVITE
Server: X-Lite release 1103a
Max-Forwards: 69
Record-Route: <sip:192.168.0.1;lr>
Record-Route: <sip:192.168.0.2;lr>
Content-Length: 0

SIP req. from : 192.168.0.2:5060 (UDP) - 5/11/2010 9:41:49 AM
SIP/2.0 180 Ringing
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7CAF9B4190022B58D;received=192.168.0.1
[Via] = SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
[From] = shawn <sip:999@192.168.0.1:5061>;tag=3543800477
[To] = <sip:123@192.168.0.2>;tag=3700807989
[Contact] = <sip:123@192.168.0.2:5061>
[Call-ID] = E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
[CSeq] = 28005 INVITE
[Server] = X-Lite release 1103a
[Max-Forwards] = 70
[Record-Route] = <sip:192.168.0.2;lr>
[Record-Route] = <sip:192.168.0.1;lr>
[Content-Length] = 0
SIP req. to : 192.168.0.1:5061 (UDP) - 5/11/2010 9:41:49 AM
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
From: shawn <sip:999@192.168.0.1:5061>;tag=3543800477
To: <sip:123@192.168.0.2>;tag=3700807989
Contact: <sip:123@192.168.0.2:5061>
Call-ID: E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
CSeq: 28005 INVITE
Server: X-Lite release 1103a
Max-Forwards: 69
Record-Route: <sip:192.168.0.1;lr>
Record-Route: <sip:192.168.0.2;lr>
Content-Length: 0

SIP req. from : 192.168.0.2:5060 (UDP) - 5/11/2010 9:41:51 AM
SIP/2.0 200 Ok
[Via] = SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7CAF9B4190022B58D;received=192.168.0.1
[Via] = SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
[From] = shawn <sip:999@192.168.0.1:5061>;tag=3543800477
[To] = <sip:123@192.168.0.2>;tag=3700807989
[Contact] = <sip:123@192.168.0.2:5061>
[Call-ID] = E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
[CSeq] = 28005 INVITE
[Server] = X-Lite release 1103a
[Max-Forwards] = 70
[Record-Route] = <sip:192.168.0.2;lr>
[Record-Route] = <sip:192.168.0.1;lr>
[Content-Type] = application/sdp
[Content-Length] = 214
v=0
o=123 1686156 1688812 IN IP4 192.168.0.2
s=X-Lite
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 3 97 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP req. to : 192.168.0.1:5061 (UDP) - 5/11/2010 9:41:51 AM
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.1:5061;received=192.168.0.1;rport=5061;branch=z9hG4bK70C0A2F29BB5481D93AA74D91927E78A
From: shawn <sip:999@192.168.0.1:5061>;tag=3543800477
To: <sip:123@192.168.0.2>;tag=3700807989
Contact: <sip:123@192.168.0.2:5061>
Call-ID: E652BE21-3F5A-45F4-977B-C404A421A8E6@192.168.0.1
CSeq: 28005 INVITE
Server: X-Lite release 1103a
Max-Forwards: 69
Record-Route: <sip:192.168.0.1;lr>
Record-Route: <sip:192.168.0.2;lr>
Content-Type: application/sdp
Content-Length: 214
v=0
o=123 1686156 1688812 IN IP4 192.168.0.2
s=X-Lite
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 3 97 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
I'm using TekSIP for both proxies in my test scenario. I think the proxy should just pass thru the Record-Route headers as they are. Right?
• permalink • reply with quote
13.05.2010 16:50:30

admin
admin
Administrator
Posts: 106 		Hi,

Can you try latest built which I've posted a couple of minutes ago? Let me know the result.

Best regards,

Yasin KAPLAN
• permalink • reply with quote
13.05.2010 16:50:30

admin
admin
Administrator
Posts: 106 		Hi,

Can you try latest built which I've posted a couple of minutes ago? Let me know the result.

Best regards,

Yasin KAPLAN
• permalink • reply with quote
17.05.2010 22:56:37

oldDude
oldDude
Posts: 16 		Seems to be working better. I'll try a few more configurations tomorrow and let you know if I see any additional problems.
• permalink • reply with quote
18.05.2010 08:56:26

admin
admin
Administrator
Posts: 106 		OK, I'll be waiting for your feedback. Thank you.
• permalink • reply with quote

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