「Cisco/CallManager Express/Debugging」の版間の差分

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159行目: 159行目:
 
  Nov  1 17:54:44.655: Parse Error: url_parseSipUrl: Received Bad Port
 
  Nov  1 17:54:44.655: Parse Error: url_parseSipUrl: Received Bad Port
 
  Nov  1 17:54:44.655: //-1/00267345B7D1/SIP/Error/sipSPIGetReqBaseCallInfo: Invalid From header
 
  Nov  1 17:54:44.655: //-1/00267345B7D1/SIP/Error/sipSPIGetReqBaseCallInfo: Invalid From header
−
==debug ccsip error==
+
==debug ccsip events==
 
SIP関連のイベントを記録する。
 
SIP関連のイベントを記録する。
  
−
  #debug ccsip error
+
  #debug ccsip events
 
   
 
   
 
  何回か発着信してみる
 
  何回か発着信してみる

2008年11月2日 (日) 03:30時点における版

CallManager Express(CME)トラブルシューティングのお供に。

ログ取りに関連するあれこれ

ログ取りの前に色々設定しておくこと

  1. 時刻を正確にしておく(NTP設定をしておく)
  2. logのバッファーを多めにとっておくか、syslogサーバーにログを垂れ流すように設定しておく。
  3. pingやtracerouteなどで、対向先(サーバーや電話機など)と正常に通信できていることを確認しておく。

電話機関連Debug

debug ephone sccp-state

SCCP(Sinny)電話機イベントの記録を取る。
電話機の上げ下ろしまで記録される。 

#debug ephone sccp-state

電話機を色々動かしてみる

#show log

Log Buffer (8192 bytes):

Nov  1 18:23:25.483: ephone-1[1]:SetCallState line 1 DN 1(-1) chan 1 ref 48 TsOffHook
Nov  1 18:23:26.595: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
Nov  1 18:23:26.603: //2593/01D9335FB800/SIP/Event/sipSPICreateRpid: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
Nov  1 18:23:26.611: ephone-1[1]:SetCallState line 1 DN 1(-1) chan 1 ref 48 TsProceed
Nov  1 18:23:26.611: ephone-1[1]:SetCallState line 1 DN 1(-1) chan 1 ref 48 TsRingOut
Nov  1 18:23:28.103: ephone-1[1]:SetCallState line 1 DN 1(1) chan 1 ref 48 TsOnHook
Nov  1 18:23:28.123: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT


SIP関連Debug

debug ccsip messages

SIPパケット(SIPのメッセージ)の記録を取る。 

#debug ccsip messages

何回か発着信してみる

#show log

(略)

Log Buffer (8192 bytes):

Nov  1 17:32:16.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:551@192.168.0.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0459620d;rport
From: "201" <sip:201@192.168.0.5>;tag=as39888bca
To: <sip:551@192.168.0.249>;tag=FF2F68-102E
Call-ID: 032ad35971809559685a9c280eaed181@192.168.0.5
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



Nov  1 17:32:16.605: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0459620d;rport
From: "201" <sip:201@192.168.0.5>;tag=as39888bca
To: <sip:551@192.168.0.249>;tag=FF2F68-102E
Date: Sat, 01 Nov 2008 17:32:16 GMT
Call-ID: 032ad35971809559685a9c280eaed181@192.168.0.5
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 BYE
Reason: Q.850;cause=16
Content-Length: 0

debug ccsip calls

SIPの発着信相手の詳細を記録する。(IPアドレス、発着信番号など) 

#debug ccsip calls

何回か発着信してみる

#show log

(略)

Log Buffer (8192 bytes):

Nov  1 17:36:06.654: //2573/650D4792B7AF/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x846343D4
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 201
Called Number            : 551
Source IP Address (Sig  ): 192.168.0.249
Destn SIP Req Addr:Port  : 192.168.0.5:5060
Destn SIP Resp Addr:Port : 192.168.0.5:5060
Destination Name         : 192.168.0.5

Nov  1 17:36:06.658: //2573/650D4792B7AF/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0
Source IP Address (Media): 192.168.0.249
Source IP Port    (Media): 19116
Destn  IP Address (Media): 192.168.0.5
Destn  IP Port    (Media): 19600
Orig Destn IP Address:Port (Media): 0.0.0.0:0

Nov  1 17:36:08.838: //2573/650D4792B7AF/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x846343D4
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 201
Called Number            : 551
Source IP Address (Sig  ): 192.168.0.249
Destn SIP Req Addr:Port  : 192.168.0.5:5060
Destn SIP Resp Addr:Port : 192.168.0.5:5060
Destination Name         : 192.168.0.5

Nov  1 17:36:08.838: //2573/650D4792B7AF/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0
Source IP Address (Media): 192.168.0.249
Source IP Port    (Media): 19116
Destn  IP Address (Media): 192.168.0.5
Destn  IP Port    (Media): 19600
Orig Destn IP Address:Port (Media): 0.0.0.0:0

debug ccsip error

SIP関連のエラーを記録する。 

#debug ccsip error

何回か発着信してみる

#show log

(略)

Log Buffer (8192 bytes):

Nov  1 17:53:11.435: Parse Error: url_display_err: Unmatched <>
Nov  1 17:53:11.439: Parse Error: url_parseSipUrl: Received Bad Port
Nov  1 17:53:11.443: //-1/xxxxxxxxxxxx/SIP/Error/sipSPICheckRequest: CheckRequest fail on method 102 error code: 9 and status: 400
Nov  1 17:53:11.443: //-1/xxxxxxxxxxxx/SIP/Error/act_idle_new_message: Failed check request
Nov  1 17:53:11.459: Parse Error: url_display_err: Unmatched <>
Nov  1 17:53:11.463: Parse Error: url_display_err: Unmatched <>
Nov  1 17:53:11.463: //-1/xxxxxxxxxxxx/SIP/Error/sippmh_cmp_tags: Parse Error in request header
Nov  1 17:53:11.463: //2582/C896C04AB7CF/SIP/Error/sipSPICheckFromToRequest:
Failed FROM/TO Request check - IGNORE IF HAIRPIN CALL
                old_from: ""501" <501>" <sip:501@211.XXX.XXX.XXX:0>;tag=as7318f14e
                old_to:   <sip:551@192.168.0.249>;tag=112E0D4-1303
                new_from: ""501" <501>" <sip:501@211.XXX.XXX.XXX:0>;tag=as7318f14e
                new_to:   <sip:551@192.168.0.249>;tag=112E0D4-1303
Nov  1 17:53:11.463: //-1/xxxxxxxxxxxx/SIP/Error/sipSPISipIncomingMsg: Invalid method for (STATE_IDLE): ACK
Nov  1 17:53:44.623: Parse Error: url_parseSipUrl: Received Bad Port
Nov  1 17:53:44.623: Parse Error: url_parseSipUrl: Received Bad Port
Nov  1 17:53:44.623: //-1/DC5E401AB7D0/SIP/Error/sipSPIGetReqBaseCallInfo: Invalid From header
Nov  1 17:54:44.651: Parse Error: url_parseSipUrl: Received Bad Port
Nov  1 17:54:44.655: Parse Error: url_parseSipUrl: Received Bad Port
Nov  1 17:54:44.655: //-1/00267345B7D1/SIP/Error/sipSPIGetReqBaseCallInfo: Invalid From header

debug ccsip events

SIP関連のイベントを記録する。 

#debug ccsip events

何回か発着信してみる

#show log
 
(略)

Log Buffer (8192 bytes):

Nov  1 18:03:12.097: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
Nov  1 18:03:12.121: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_ALERTING
Nov  1 18:03:13.145: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_MEDIA_CHANGED
Nov  1 18:03:13.153: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_CONNECT
Nov  1 18:03:15.221: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

外部リンク

Verifying and Troubleshooting SIP Features
CME SIP関連機能のデバッグ方法(英語)
「https://www.voip-info.jp/index.php?title=Cisco/CallManager_Express/Debugging&oldid=2865」から取得