[Asterisk-Users] SIP/2.0 487 Request Cancelled

[Jess Magnaye]

Here's my sip debug output.  anybody knows why Cisco sent * is CANCEL msg? Can someone tell me what ATA version are they using? Maybe this is also another issue.. I am using v2.16.

This is using G711ulaw.



Sip read: > 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e
From: "Jess" <sip:6882332 at mydomain.com>;tag=as6818ebfb
To: <sip:0119614960203 at provider>;tag=243881BC-1D1
Date: Fri, 09 Jan 2004 17:56:09 GMT
Call-ID: 27907464513ff22b6c7a203c51b3a666 at asterisk
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


10 headers, 0 lines
Sip read: > 
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e
From: "Jess" <sip:6882332 at mydomain.com>;tag=as6818ebfb
To: <sip:0119614960203 at provider>
Date: Fri, 09 Jan 2004 17:56:09 GMT
Call-ID: 27907464513ff22b6c7a203c51b3a666 at asterisk
Content-Length: 0
CSeq: 102 CANCEL


8 headers, 0 lines
Sip read: > 
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e
From: "Jess" <sip:6882332 at mydomain.com>;tag=as6818ebfb
To: <sip:0119614960203 at provider>;tag=243881BC-1D1
Date: Fri, 09 Jan 2004 17:56:09 GMT
Call-ID: 27907464513ff22b6c7a203c51b3a666 at asterisk
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


--------- FROM CISCO -------------

Jan  9 17:20:39: Sent: 
SIP/2.0 100 Trying          
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2          
From: "Jess" <sip:6882332 at mydomain.com>;tag=as7f1f229e          
To: <sip:260749614960203 at provider>;tag=241801E4-2525          
Date: Fri, 09 Jan 2004 17:20:39 GMT          
Call-ID: 0d696fb61d406bc653cceeef5596c54d at asterisk     
Server: Cisco-SIPGateway/IOS-12.x          
CSeq: 102 INVITE          
Allow-Events: telephone-event          
Content-Length: 0          
          
          
          
Jan  9 17:20:39:  Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
Jan  9 17:20:39: ccsip_report_digit_control: enable=0: 
Jan  9 17:20:39:  ccsip_report_digit_control: disabled.
Jan  9 17:20:39: *****CCB found in UAS Request table. ccb=0x642D4630
Jan  9 17:20:39: CCSIP-SPI-CONTROL:  act_recdinvite_new_message
Jan  9 17:20:39: CCSIP-SPI-CONTROL:  Clock Time Zone is UTC, same as GMT: Using GMT
Jan  9 17:20:39: sip_stats_method
Jan  9 17:20:39: ccsip_set_release_source_for_peer:ownCallId[1132803], src[2]
          
Jan  9 17:20:39: 0x642D4630 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
Jan  9 17:20:39:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
Jan  9 17:20:39: sip_stats_status_code
Jan  9 17:20:39: CCSIP-SPI-CONTROL:  sipSPISendInviteResponse
Jan  9 17:20:39:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
Jan  9 17:20:39: sip_stats_status_code
Jan  9 17:20:39: CCSIP-SPI-CONTROL:  sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for incoming call
Jan  9 17:20:39: 0x642D4630 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
Jan  9 17:20:39: Ignoring unexpected event 11 (SIPSPI_EV_CC_CALL_PROCEEDING) in state 8 (STATE_DISCONNECTING) substate 0 (SUBSTATE_NONE)
Jan  9 17:20:39:  Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Jan  9 17:20:39: Sent: 
SIP/2.0 200 OK          
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2          
From: "Jess" <sip:6882332 at mydomain.com>;tag=as7f1f229e          
To: <sip:260749614960203 at provider>          
Date: Fri, 09 Jan 2004 17:20:39 GMT          
Call-ID: 0d696fb61d406bc653cceeef5596c54d at asterisk    
Content-Length: 0          
CSeq: 102 CANCEL          
          
          
          
Jan  9 17:20:39: Sent: 
SIP/2.0 487 Request Cancelled          
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK7db389e2          
From: "Jess" <sip:6882332 at mydomain.com>;tag=as7f1f229e          
To: <sip:260749614960203 at provider>;tag=241801E4-2525          
Date: Fri, 09 Jan 2004 17:20:39 GMT          
Call-ID: 0d696fb61d406bc653cceeef5596c54d at asterisk        
Server: Cisco-SIPGateway/IOS-12.x          
CSeq: 102 INVITE          
Allow-Events: telephone-event          
Content-Length: 0  
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