두 엔드포인트 간의 SIP 통화 문제 해결

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업데이트:2006년 4월 27일
문서 ID:69467

편견 없는 언어

본 제품에 대한 문서 세트는 편견 없는 언어를 사용하기 위해 노력합니다. 본 설명서 세트의 목적상, 편견 없는 언어는 나이, 장애, 성별, 인종 정체성, 민족 정체성, 성적 지향성, 사회 경제적 지위 및 교차성에 기초한 차별을 의미하지 않는 언어로 정의됩니다. 제품 소프트웨어의 사용자 인터페이스에서 하드코딩된 언어, RFP 설명서에 기초한 언어 또는 참조된 서드파티 제품에서 사용하는 언어로 인해 설명서에 예외가 있을 수 있습니다. 시스코에서 어떤 방식으로 포용적인 언어를 사용하고 있는지 자세히 알아보세요.

이 번역에 관하여

Cisco는 전 세계 사용자에게 다양한 언어로 지원 콘텐츠를 제공하기 위해 기계 번역 기술과 수작업 번역을 병행하여 이 문서를 번역했습니다. 아무리 품질이 높은 기계 번역이라도 전문 번역가의 번역 결과물만큼 정확하지는 않습니다. Cisco Systems, Inc.는 이 같은 번역에 대해 어떠한 책임도 지지 않으며 항상 원본 영문 문서(링크 제공됨)를 참조할 것을 권장합니다.

소개

이 문서에서는 두 게이트웨이 간에 SIP(Session Initiation Protocol) 통화가 발생하는 방법을 보여 주기 위해 두 팩스 장치의 샘플 컨피그레이션을 제공합니다. 이 문서에서는 SIP 통화 실패 문제 해결을 위한 debug ccsip messages 명령의 출력에 대한 설명을 제공합니다.

사전 요구 사항

요구 사항

이 문서에 대한 특정 요건이 없습니다.

사용되는 구성 요소

이 문서의 정보는 다음 소프트웨어 및 하드웨어 버전을 기반으로 합니다.

  • 팩스 2대

  • Cisco IOS® Software 릴리스 12.4(4)T1을 실행하는 VG224

  • Cisco IOS Software 릴리스 12.3(11)T8을 실행하는 Cisco 3745 라우터

이 문서의 정보는 특정 랩 환경의 디바이스를 토대로 작성되었습니다. 이 문서에 사용된 모든 디바이스는 초기화된(기본) 컨피그레이션으로 시작되었습니다. 현재 네트워크가 작동 중인 경우, 모든 명령어의 잠재적인 영향을 미리 숙지하시기 바랍니다.

표기 규칙

문서 규칙에 대한 자세한 내용은 Cisco 기술 팁 표기 규칙을 참고하십시오.

구성

이 섹션에는 이 문서에서 설명하는 기능을 구성하기 위한 정보가 표시됩니다.

참고: 명령 조회 도구(등록된 고객만 해당)를 사용하여 이 문서에 사용된 명령에 대한 자세한 내용을 확인하십시오.

네트워크 다이어그램

이 문서에서는 다음 네트워크 설정을 사용합니다.

sip-calls-1.gif

구성

이 문서에서는 다음 구성을 사용합니다.

VG224 
vg224#show run
Building configuration...
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
voice-port 2/0
 idle-voltage low
!
dial-peer voice 1 pots
<fax machine connected to this port>
 destination-pattern 9000
 port 2/0
!
dial-peer voice 100 voip
 destination-pattern 8000
 no modem passthrough
 session protocol sipv2
 session target ipv4:172.16.184.83
 incoming called-number .
 codec g711ulaw
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
!

Cisco 3745 
HTTS-VRK1-3745-1#show run
Building configuration...
!
voice service voip
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
!
voice-port 4/1/0
!
!
dial-peer voice 9000 voip
 destination-pattern 9000
 session protocol sipv2
 session target ipv4:172.16.13.87
 incoming called-number .
 codec g711ulaw
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!
dial-peer voice 9 pots
 destination-pattern 8000
 fax rate voice
 port 4/1/0
 forward-digits all

다음을 확인합니다.

현재 이 구성에 대해 사용 가능한 확인 절차가 없습니다.

문제 해결

이 섹션에서는 컨피그레이션 문제를 해결할 수 있습니다.

Output Interpreter 도구(등록된 고객만 해당)(OIT)는 특정 show 명령을 지원합니다. OIT를 사용하여 show 명령 출력의 분석을 봅니다.

참고: debug 명령을 사용하기 전에 디버그 명령에 대한 중요 정보를 참조하십시오.

다음은 debug ccsip messages 명령의 출력입니다. 


!--- This is the first invite message sent out !--- to the terminating SIP gateway. !--- This is similar to a setup message in H.323 or Q.931.

*Mar 1 00:33:42.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:8000@172.16.184.83:5060 SIP/2.0

!--- 8000 is the DN of the call, 172.16.184.83 is !--- the IP address of the remote gateway, and !--- 5060 is the port the SIP works on. !--- This configuration uses SIP version 2.0.

Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF

!--- The VIA field is used for devices in the patch that !--- need to be aware of the call. !--- In this case, there are no SIP devices in between the two gateways.

Remote-Party-ID: <sip:9000@172.16.13.87>;party=calling;screen=no;privacy=off
 !--- The DN and URI of the remote SIP device that is called.

From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>
Date: Fri, 01 Mar 2002 00:33:42 GMT

!--- The time that the invite is sent out

Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87

!--- The call ID is unique for every call. !--- This ID is used to identify a particular call !--- in a busy router.

Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3481906499-736235990-2149183265-3714191467
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE

!--- The sequence number for each transaction.
 
Max-Forwards: 70
Timestamp: 1014942822
Contact: <sip:9000@172.16.13.87:5060>

!--- This is the address used to reach the calling party on the return path.

Expires: 180

!--- This message expires in 180 seconds.

Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
 !--- The Session Descriptor Protocol (SDP) version is zero. !--- This is different from the SIP version used !--- in this example configuration.

o=CiscoSystemsSIP-GW-UserAgent 1715 2724 IN IP4 172.16.13.87

!--- The owner of the device that created the call. !--- This is sometimes referred to as organization.

s=SIP Call

!--- The name given to this particular SIP call. This is the description.

c=IN IP4 172.16.13.87

!--- Connection information. Usually includes the IP address of !--- the originating device. It is an optional field.

t=0 0
m=audio 18080 RTP/AVP 0 19

!--- This is the media information. In this case, !--- 18080 is used as the UDP port for RTP.

c=IN IP4 172.16.13.87
a=rtpmap:0 PCMU/8000

!--- This is the media attributes. Notice the 0 and 19 in !--- the media field. These are the !--- attributes that go with that. PCMU/8000 is G711ulaw.

a=rtpmap:19 CN/8000
a=ptime:20

!--- A packetization period of 20 ms.



!--- In this output, invite, SDP is not a required parameter. !--- But in this case you see that SDP sent out. !--- SDP carries information about capabilities. !--- No information about fax capabilities are !--- exchanged in the beginning because it is only a voice !--- call until you hear fax tones from the terminating fax machine.


*Mar 1 00:33:43.203: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Content-Length: 0


!--- The terminating machine sets up an analog !--- connection to the fax machine, and while it waits, !--- it sends a "trying" message. This stops the !--- originating gateway from sending another invite.


*Mar 1 00:33:43.207: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Require: 100rel
RSeq: 3696
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0

!--- This is a different UDP port for the reverse direction.

c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- A "progress" indicator tells you that the remote gateway sent a connect !--- and the fax machine is ringing at this time. !--- Note that the To and From headers do not change despite !--- the fact that the message comes in the opposite direction.


*Mar 1 00:33:43.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Require: 100rel
RSeq: 3696
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0
c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- A provisional ack to the progress message.


*Mar 1 00:33:43.251: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
PRACK sip:8000@172.16.184.83:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Fri, 01 Mar 2002 00:33:42 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
CSeq: 102 PRACK
RAck: 3696 101 INVITE
Max-Forwards: 70
Content-Length: 0


!--- This is an OK for the PRACK. You can tell this from the Cseq header.


*Mar 1 00:33:44.031: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:37 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 PRACK
Content-Length: 0


!--- An OK is received, which is mandatory for an invite. !--- The OK has information on the accepted media parameters in the SDP.


*Mar 1 00:33:49.431: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:37 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0
c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- The ack for the OK.


*Mar 1 00:33:49.443: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:8000@172.16.184.83:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKD1A5C
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Fri, 01 Mar 2002 00:33:42 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


!--- At this point, the terminating gateway hears fax tones and determines it !--- has to switch the codec to a !--- fax codec and sends a re-invite. The re-invite contains !--- information about the new media !--- parameters that the terminating gateway wants to change to. 

*Mar 1 00:33:55.247: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:43:49 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 3481906499-736235990-2149183265-3714191467
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1141170229
Contact: <sip:8000@172.16.184.83:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 399

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2736 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=image 18304 udptl t38
c=IN IP4 172.16.184.83
a=T38FaxVersion:0
a=T38MaxBitRate:14400

!--- The maximum bit rate that is supported by the terminating gateway.

a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy

!--- UDP redundancy is enabled.



!--- A trying message is sent and an !--- attempt is made to determine if T.38 fax-relay is supported.


*Mar 1 00:33:55.275: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:33:55 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Remote-Party-ID: <sip:9000@172.16.13.87>;party=called;screen=no;privacy=off
Content-Length: 0


!--- The OK to the re-invite that specifies that you can !--- do T.38 fax-relay. The same UDP port is retained.

 
*Mar 1 00:33:55.275: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:33:55 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:9000@172.16.13.87>;party=called;screen=no;privacy=off
Contact: <sip:9000@172.16.13.87:5060>
Content-Type: application/sdp
Content-Length: 157

v=0
o=CiscoSystemsSIP-GW-UserAgent 1715 2725 IN IP4 172.16.13.87
s=SIP Call
c=IN IP4 172.16.13.87
t=0 0
m=image 18080 udptl t38
c=IN IP4 172.16.13.87


!--- The ack to the OK is received. At this point, fax transmission occurs. 

*Mar 1 00:33:55.719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1B21D0
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:43:49 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


!--- Once the fax transmission is completed, !--- the BYE is received. The BYE is similar to a !--- release message in Q.931.


*Mar 1 00:34:45.515: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
BYE sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1E1E51
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:44:38 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1141170279
CSeq: 103 BYE
Reason: Q.850;cause=16

!--- Cause code 16 is a normal disconnect cause.

Content-Length: 0

!--- There should be an OK to every message.

*Mar 1 00:34:45.535: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1E1E51
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:34:45 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
Timestamp: 1141170279
CSeq: 103 BYE
Reason: Q.850;cause=16
Content-Length: 0
 

More information about the attributes:

Session description
v= (protocol version)
o= (owner/creator and session identifier).
s= (session name) 
i=* (session information)
u=* (URI of description)
e=* (email address)
p=* (phone number)
c=* (connection information - not required if included in all media)
b=* (bandwidth information)
z=* (time zone adjustments)
k=* (encryption key)
a=* (zero or more session attribute lines)

Time description
t= (time the session is active)
r=* (zero or more repeat times)

Media description
m= (media name and transport address)
i=* (media title)
c=* (connection information - optional if included at session-level)
b=* (bandwidth information)
k=* (encryption key)
a=* (zero or more media attribute lines)
* indicated optional item.

Basic Requests
INVITE: request from a UAC to initiate a session
ACK: confirms receipt of a final response to INVITE
BYE: sent by either side to end a session
CANCEL: sent to end a call not yet connected
UPDATE: Updates offer for not-yet-established sessions.
REGISTER: UA registers with Registrar Server
NOTIFY: notifies that an event has occurred
REFER: the mechanism to initiate a session transfer
INFO: a means of carrying ?data? in a message body


SIP responses:

1xx: Provisional ? request received, continuing to process the request 
2xx: Success - action was successfully received, understood, and accepted 
3xx: Redirection - further action needs to be taken in order to complete the request 
4xx: Client Error - the request contains bad syntax or cannot be 
fulfilled at this server 
5xx: Server Error - the server failed to fulfill an apparently valid request 
6xx: Global Failure - the request cannot be fulfilled at any server

관련 정보

개정 이력

개정 게시 날짜 의견
1.0
27-Apr-2006
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