My won´t smartwatch 2 calls

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  Hi all

  I have a problem with Cisco SPA112 ATA. I can make calls without problem. When someone calls the ATA configured number, analog phone attached to rings of the ATA, but when I answer, I don't hear anything. The calling party infiltrates the waiting tone.

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  Kind regards

  Tom

  NEW_CALL_STATE(), dial 0: former State = CC_CST_IDLE, new State CC_CST_RINGING
  CID_initGen 8
  [0]:CID_initGen() Cid > won 0 delay 2200 phone 501 501 * name *.
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  RTP_nextMediaPort(), rc = 16440
  AUD_allocCallObj() call (0x1b1d40)
  [0:0] DIAL the ALLOC AUD (port = 16440)
  +++ SIP_lineCcCmdProc CC_CMD_ACCEPT AUD_startRtpRx
  [AUD] AUD_startRtpRx (0x1b1d40) cover 0
  Local loop mode: No. Type: No.
  Remote loopback mode: No. Type none.
  Configuration of the RTP channels: udp_no_checksum 0, sysmmetric_rtp 0, tos 0xb8, cos mlb 6, 0.
  uchConnectEpToNode(), connection VoIP EP 0 node 0
  uchEnableEchoCan(), cover 0 EP 2 turn on
  uchEnableModemCall() Modem call state (0) no change
  UCH sync parameter hold off time is 70
  Succeed together QoS
  uchSetGTD(), disable GTD 1 FXS
  State GTD uchSetGTD() not change
  uchSetDTMFMute(), DISABLE
  cordless_start_rtp(), chan: 0 ip: (null) remote port: 0 local: 16440 rx: 1 ipt:0 ptime:30 bInMdmPasstru:0
  From RTP only Rx.
  Decision-making related to 16440 port 14.
  Remote IP/port: 0.0.0.0:0
  SDP codec list (internal pt): 0 18 8 134 136
  List of load RX: PCMU/8000 (0) G.729/8000 (18) PCMA/8000 (8) NSE/8000 (100) encaprtp/8000 (112)
  Set RTP_SESSION_OPT_DTMF
  VAD = 0
  RTP configuration:
  audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opt 0 x 0
  Codec: length 30, 101, 101 tx_pt_event rx_pt_event, tx_pt 0
  RX [0] 0 PCMU/8000, rx [1] 18 G.729/8000, rx [2] 8 PCMA/8000
  RX [3] 100 NSE/8000, rx [4] 112 encaprtp/8000, rx [5] - 1
  Stem: 180ms max, min 60ms, adapt 1
  RTP channel 0 is blank: 1.
  # rtp seq number is 668
  RTP session 0 has begun
  [AUD] RTP Rx upward
  Pol CID:OnHookTx
  [0] CID CID_ST_POLREV_POST_DELAY
  uchPlayDTASTone(), playuchSetMute(), SWITCH, ret = 0
  [0] CID CID_ST_CAS
  [0] CID CID_ST_CAS_POST_DELAY
  uchDisplayCIDFSK(), EP 2 0 95 buflen cover costs 60 generals SZ_MAX_USERDATA 200 won 0
  uchDisplayCIDFSK(), FSK Caller ID standard is 0 (bell 202)
  uchDisplayCIDFSK(), MarkFreq SeizeFreq 0 x 16 0xc
  [0] CID Start DTMF/FSK, CID_ST_ACTIVE
  uchAppCb(), 49 received EP 2 0 cover event
  receive CH_ASYNC_CIT_TRANSMITTED
  [0] CID CID: FACT
  [0] CID CID_ST_ACTIVE_POST_DELAY
  [0] CID CID_ST_IDLE
  CID:ring now
  CC_eventProc(), event: CC_EV_CID_DONE (0 x 51), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 81 vid 0 by 0 x 0 par2 0x0
  SLIC_startRing of State 0 ts 0x1af54con 2000 stop 4000 len 60000
  [0] ring cad event 0-pol 0
  uchSetMute(), DISABLE, ret = 0
  [0] ring cad event 1-pol 0

  SIP_tsCreateClient(), 1671, uiTmrF = 1600, SIP_TMR_F_INIT = 1600
  SIP_tsClientEventProc(Event: 28) State = 1
  SIP_regTsEventProc(Event: 28)
  SIP_regTsEventProc(Event: 32)
  Detached TS
  SIP_tsClientEventProc(Event: 3) status = 3
  [0] ring cad event 0-pol 0
  [0] ring cad event 1-pol 0
  SIP_tsClientEventProc(Event: 9) status = 3
  [0] off hook
  CC_eventProc(), event: CC_EV_USR_OFFHOOK (0x2), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 2 vid 0 by 0 x 0 par2 0x0
  sysstatus_set_led_status_payton(), led_id: 1, statusCode:7, systemEvent: 0 x 100095
  callEventProcTable [5] is cepRingingProc
  cepRingingProc (lid = 0, call = 0x17f9f4, event = 11 (CC_EV_USR_ANSWER), by = 0, par2 = (Nile))
  NEW_CALL_STATE(), dial 0: former State = CC_CST_RINGING, new State CC_CST_ANSWERING
  SLIC_stopRing
  [0] ring cad event 2-pol 0
  SLIC_stopRing
  SLIC_stopTone
  uchStopVoipTone(), stop Voip your EP 3
  Start the timer G and H (e = 22)
  TMR shot G
  TMR shot G
  TMR shot G
  SIP_sessTsEventProc(Event:27)
  Departure TmrJ
  SIP_sessDlgEventProc: event: 45 (SIP_EV_DLG_BYED), ucState: 0
  Entered SIP_releaseAudioResources() #!
  Statistics of calls asking...
  Stats of RTP TX updated for channel 0
  Stats of RTP RX updated for channel 0
  Call the updated statistics.
  AUD_releaseCallObj() call (0x1b1d40)
  [AUD] AUD_stopRtpTx (0x1b1d40)
  cordless_stop_rtp_tx(), channel 0.
  Channel RTP not in Tx. Nothing can stop!
  Channel RTP not in Tx. Nothing can stop!
  [AUD] RTP low Tx
  [AUD] AUD_stopRtpRx (0x1b1d40)
  cordless_stop_rtp_rx(), channel 0.
  RTP channel 0 ranging from Rx to idle.
  RTP configuration:
  audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opt 0 x 0
  Codec: length 30, 101, 101 tx_pt_event rx_pt_event, tx_pt 0
  RX [0] 0 PCMU/8000, rx [1] 18 G.729/8000, rx [2] 8 PCMA/8000
  RX [3] 100 NSE/8000, rx [4] 112 encaprtp/8000, rx [5] - 1
  Stem: 180ms max, min 60ms, adapt 1
  RTP channel 0 is now inactive.
  [AUD] RTP down
  [AUD] AUD_releaseRtp (0x1b1d40)
  cordless_stop_rtp(), freeing the channel RTP: 0
  cordless_stop_rtp(), RTP session 0 arrested succussfully
  uchRelChanAndEP (0, 3)
  uchDisconnectEpFromNode(), disconnection VoIP EP 0 node 0
  [AUD] RTP channel released
  [0:0] call of Rel AUD
  SIP_releaseAudioResources(), CC_lineIsIdle (0) = 0, gAudLine [0] .bIvr = 0, AUD_relUchNode?
  Output SIP_releaseAudioResources() #!
  CC_eventProc(), event: CC_EV_SIG_CALL_ENDED (0 x 34), cover: 0, rating: 6, par2: (none)
  AUD_ccEventProc: event 52 vid 0 by 0 x 6 par2 0x0
  callEventProcTable [7] is cepAnsweringProc
  cepAnsweringProc (lid = 0, call = 0x17f9f4, event = 52 (CC_EV_SIG_CALL_ENDED), by = 6, par2 = (Nile))
  CC: finished
  NEW_CALL_STATE(), dial 0: former State = CC_CST_ANSWERING, new State CC_CST_INVALID
  Statr CPC CC_EV_TMR_INVALID timer
  SLIC_stopRing
  SLIC_stopRing
  SLIC_stopTone
  # 1248 RTP_SEQ_NUM_EVT
  [0] on the hook
  CC_eventProc(), event: CC_EV_USR_ONHOOK (0x1), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 1 vid 0 by 0 x 0 par2 0x0
  sysstatus_set_led_status_payton(), led_id: 1, statusCode:1, systemEvent: 0 x 100093
  callEventProcTable [6] is cepInvalidProc
  cepInvalidProc (lid = 0, call = 0x17f9f4, event = 1 (CC_EV_USR_ONHOOK), by = 0, par2 = (Nile))
  callEventProcTable [0] is cepIdleProc
  cepIdleProc (lid = 0, call = 0x17fc14, event = 1 (CC_EV_USR_ONHOOK), by = 0, par2 = (Nile))
  cepIdleProc(), lid = 0
  [IVR_eventProc] evt 1 lid 0
  callEventProcTable [6] is cepInvalidProc
  cepInvalidProc (lid = 0, call = 0x17f9f4, event = 10 (CC_EV_USR_ENDCALL), by = 0, par2 = (Nile))
  NEW_CALL_STATE(), dial 0: former State = CC_CST_INVALID, new State CC_CST_IDLE
  [AUD] Release knot UCH for AUD_LINE 0.
  uchDisableNode(), node released 0 ret = 0
  [AUD] Node UCH released 0.
  SLIC_stopRing
  SLIC_stopTone
  Succeed together QoS
  TMR shot G

  SIP_tsCreateClient(), 1671, uiTmrF = 1600, SIP_TMR_F_INIT = 1600
  SIP_tsClientEventProc(Event: 28) State = 1
  SIP_regTsEventProc(Event: 28)
  SIP_regTsEventProc(Event: 32)
  Detached TS
  SIP_tsClientEventProc(Event: 3) status = 3
  TMR shot G
  TP Analyzer error: 34
  SIP_tsClientEventProc(Event: 9) status = 3
  TMR shot G
  TMR shot G
  TMR shot G

  16:31:23 [0]Off Hook16:31:23 NEW_CALL_STATE(), call 0: old state = CC_CST_RINGING, new state CC_CST_ANSWERING16:31:49 SIP_sessDlgEventProc: event: 45(SIP_EV_DLG_BYED), ucState: 0

  Phone go off the hook. Setup of the call must be made here. I guess that the anti-terrorism Act has sent 200 OK "response to confirmation by peers. It has not been recognized, call was BYEd peer (after 20 sec timeout).

  The disconnection happens indeed on the remote side but after 'no answer'.

  No way to have two the same call disconnection. Mobile phone received 'no response' for a reason. And your ATA did not sign-out request (well, I guess with any SIP connects available). The disconnection was initiated by an outside entity to you. Yes, it can be caused by some session parameters used by the anti-terrorism Act, but with no available SIP newspaper, we have little chance to guess the true cause. My best advice is (assuming that you are unable to capture the SIP packets now) - call your support operator for help. I suspect that the next PBX (from your side) launches the disconnection so logs of your operator should disclose the cause in its entirety.

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