FXO end ok-can talk BUT Dial tone not receiving at fxs side

craaj
Joined: Tue 11 of Jun, 2013

FXO end ok-can talk BUT Dial tone not receiving at fxs side

Posted:Wed 12 of Jun, 2013 (07:42 UTC)
Scenerio-- Two Mp FXo and Fxs were running smoothly; After a power fault we change device at one end and at other end we
reset the device as password not known, but we had INI config saved. Now

Between two p2p site, from FXO connected to pax, we(remote users) can talk and listen seemlessly but at fxs
end located in another location; we are not sensing dial tone not also able to communicate dialing no, so captured log file
which basically states in the last lines "[ERROR] SIPCall can't handle Disconnect event in state Cancelling"
Pls help me voip/sip experts as Iam novish in this field. If you need INI files pls tell my id is choudhury.rajesh1@gmail.com
I can also call if someone of you come forward to help me out.

Log is Activated  1d:16h:31m:53s (   lgr_psbrdex)(2588      )  recv <-- OFF_HOOK Ch:3
1d:16h:31m:53s (      lgr_flow)(2589      )  #3:OFF_HOOK_EV
1d:16h:31m:53s (      lgr_flow)(2590      )  |       #3:OFF_HOOK_EV
1d:16h:31m:53s (   lgr_psbrdif)(2591      )  UpdateChannelParams, Channel 3 

1d:16h:31m:53s (   lgr_psbrdif)(2592      )  #3:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=1, VxxTranType=2, VoiceVol= 0, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=0, ECNlpMode=0, DJBufMinDelay=10, DJBufOptFac=10)
1d:16h:31m:53s (      lgr_flow)(2593      )  |       #3:NEW_CALL_EV (send)  : (UnKnown)
1d:16h:31m:53s (      lgr_flow)(2594      )  |       |       #3:NEW_CALL_EV:(UnKnown)
1d:16h:31m:53s (  lgr_stk_mngr)(2595      )  Resource StackSession <#3> Allocated
1d:16h:31m:53s (      lgr_flow)(2596      )  |       |       #3:Call changing states from:IdleState to:NewCallState_Tel2IP
1d:16h:31m:53s (      lgr_flow)(2597      )  |       |       |       #3:NEW_CALL_EV(UNKnown)
1d:16h:31m:53s (      lgr_flow)(2598      )  |       (to 54)
1d:16h:31m:53s (      lgr_flow)(2599      )  |       #3:SETUP (send)  : (UnKnown)
1d:16h:31m:53s (      lgr_flow)(2600      )  |       |       #3:SETUP (TO:54, FROM:64):(UnKnown)
1d:16h:31m:53s (      lgr_flow)(2601      )  |       |       #3:Call changing states from:NewCallState_Tel2IP to:InitiatedState_Tel2IP
1d:16h:31m:53s (   lgr_stk_ses)(2602      )  SIPStackSession::GetPrefixSearchFormat, ProxyWorking = 0, ReplaceReason = 0, SENDINVITE2PROXY = 0
1d:16h:31m:53s ( lgr_profiling)(2603      )  <Call 3> Profiled<Tel=0,Ip=0>: TelCoderGrId=0 IpCoderGrId=0 JBMinDel=10 JBOptF=10 EEarlyM=0 FaxTM=0 IPDS=46 IsFaxU=0 PI2IP=-1 SigIPDF=40 CNGMode=0 DTMFUsed=0 NSEMode=0 PlayRBTone2IP=0 RBUdpPort=0 RTPRD=0 SCE=0 VxxTT=2 DTMFVol=20 ECE=1 ECurDis=0 EDigDel=0 ERevP=0 FHPer=400 InG=32 MWIA=0 MWID=0 VVol=32
1d:16h:31m:53s (   lgr_stk_ses)(2604      )  FindIpDestination: rmRc:0 (OK) IpconnHndl:-1 DstPrefix:54 DstIp:-1072658546
1d:16h:31m:53s (      lgr_flow)(2605      )  |       |       |       #3:SETUP(UNKnown)
1d:16h:31m:53s (     sip_stack)(2606      )  new AcSIPCallAPI created - #3
1d:16h:31m:53s (      lgr_flow)(2607      )  |       | new GetNewSIPCall created - #7
1d:16h:31m:53s (     sip_stack)(2608      )  SIPSDPSession#3 - Changing state from SIP_MEDIA_IDLE to SIP_MEDIA_OFFERING
1d:16h:31m:53s (      lgr_flow)(2609      )  |       |(SIPTU#7)SETUP_REQ State:Idle()
1d:16h:31m:53s (     sip_stack)(2610      )  SIPCall(#7) changes state from Idle to Inviting
1d:16h:31m:53s (      lgr_flow)(2611      )  ---- Outgoing SIP Message to 192.16.135.142:5060 ----
1d:16h:31m:53s INVITE sip:54@192.16.135.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.134.198;branch=z9hG4bKac1913671489
Max-Forwards: 70
From: <sip:64@172.16.134.198>;tag=1c1913667840
To: <sip:54@192.16.135.142;user=phone>
Call-ID: 1913667503162013163153@172.16.134.198
CSeq: 1 INVITE
Contact: <sip:64@172.16.134.198>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-114 FXS/v.5.00A.024
Content-Type: application/sdp
Content-Length: 289

v=0
o=AudiocodesGW 1913655809 1913655687 IN IP4 172.16.134.198
s=Phone-Call
c=IN IP4 172.16.134.198
t=0 0
m=audio 6030 RTP/AVP 4 96
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:60
a=sendrecv
a=rtcp:6031 IN IP4 172.16.134.198

1d:16h:31m:53s (     sip_stack)(2613      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 6 Dest: c010878e:5060
1d:16h:31m:53s (     sip_stack)(2614      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:31m:54s (     sip_stack)(2615      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 5 Dest: c010878e:5060
1d:16h:31m:54s (     sip_stack)(2616      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:31m:55s (     sip_stack)(2617      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 4 Dest: c010878e:5060
1d:16h:31m:55s (     sip_stack)(2618      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:31m:57s (     sip_stack)(2619      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 3 Dest: c010878e:5060
1d:16h:31m:57s (     sip_stack)(2620      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:32m:1s (     sip_stack)(2621      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 2 Dest: c010878e:5060
1d:16h:32m:1s (     sip_stack)(2622      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:32m:9s (     sip_stack)(2623      )  
UdpRtxMngr::Transmit 1 INVITE Rtx Left: 1 Dest: c010878e:5060
1d:16h:32m:9s (     sip_stack)(2624      )  SIPTransaction::ResendLastMessage - Resending last message
1d:16h:32m:9s (   lgr_psbrdex)(2625      )  recv <-- ON_HOOK Ch:3
1d:16h:32m:9s (      lgr_flow)(2626      )  #3:ON_HOOK_EV
1d:16h:32m:9s (      lgr_flow)(2627      )  |       #3:ON_HOOK_EV
1d:16h:32m:9s (      lgr_flow)(2628      )  |       #3:RELEASE (send) GWAPP_NORMAL_CALL_CLEAR : (1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (      lgr_flow)(2629      )  |       |       #3:RELEASE:(1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (      lgr_flow)(2630      )  |       |       #3:Call changing states from:InitiatedState_Tel2IP to:DisconnectingState
1d:16h:32m:9s (      lgr_flow)(2631      )  |       |       #3:RELEASE_ACK:(1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (      lgr_flow)(2632      )  |       |       |       #3:RELEASE(1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (      lgr_flow)(2633      )  |       |(SIPTU#7)DISCONNECT_REQ State:Inviting(1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (     sip_stack)(2634      )  SIPCall(#7) changes state from Inviting to Cancelling
1d:16h:32m:9s (     sip_stack)(2635      )  UdpRtxMngr::Remove 1 INVITE
1d:16h:32m:9s (   lgr_stk_ses)(2636      )  <SESSION #3> SendToCall - event: RELEASE_ACK  m_Call = 32054032
1d:16h:32m:9s (      lgr_flow)(2637      )  |       |       #3:RELEASE_ACK:(1913667503162013163153@172.16.134.198)
1d:16h:32m:9s (     sip_stack)(2638      )  AcSIPStackAPI::FreeCallAPI - #3
1d:16h:32m:9s (     sip_stack)(2639      )  Setting ApplicationCall of AcSIPCall 31677504 to NULL
1d:16h:32m:9s (  lgr_stk_mngr)(2640      )  Resource StackSession <#3> Deleted
1d:16h:32m:9s (   lgr_psbrdif)(2641      )  #3:PSOSBoardInterface::StopPlayTone- Called
1d:16h:32m:25s (      lgr_flow)(2642      )  |       |(SIPTU#7)DISCONNECT_REQ State:Cancelling(1913667503162013163153@172.16.134.198)
1d:16h:32m:25s (     sip_stack)(2643      ) !! [ERROR] SIPCall can't handle Disconnect event in state Cancelling
1d:16h:32m:25s (      lgr_flow)(2644      )  |       | TransactionUserMngr::ReturnSIPCall - #7
1d:16h:32m:25s (     sip_stack)(2645      )  SIPCall(#7) changes state from Cancelling to Idle