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Outbound Dialler IVR based campaign not working. (IP shows 0.0.0.0 on VG).

Ritesh Desai
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Hi friends,

Need your support in guiding for solution. I've setup Outbound Dialler IVR Campaign based in Predictive mode. Call is initiated & connected problem is

Config:

  • PG2A configured for Dialer and MR PG.
  • CT, DN, SG, Route & Agent created for OB script and assigned to it.

 

ICM Script flow:

Start - DN - Send to VRU - Run Ext Script(PM_Welcome) - Queue to SG - Run Ext Script(PM_Thankyou).

 

Issue:

  • I don't hear Welcome and thank you prompt.
  • No call progress seen when ICM script in monitor mode.
  • Router log viewer does not show any logs for connected call or failure one.

 

Doubts:

  • Should I've to create DN for Answering Machine Detection and assign the DN in respective OB Campaign.
  • In ICM Admin Script, when "outbound control" variable set to PREDICTIVE_ONLY, is it right syntax or wrong.

 

TIA,

 

 

regards,

Ritesh Desai

 

 

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai
33 Replies 33

Hi Chintan,

Good to see you men!

 

No, I don't see success path from TR.

Call is disconnecting before hitting ICM Script. I have debugged the cause for error. I have attached Logs from VG cum VXML, PG1A-PIM1 (CUCM), PG1A-PIM2 (VRU), RTR, PG2A PIM1 (Dialer).

 

I've found negatives on VG, BADialer, PG1A(Dialer).

I suspect communication crack between Sip Dialer and MR-PG.

Error:

1. MR-PG's Peripheral Dialer naming was not matching with Dialer name while defined during setup causing for communication break.

2. In VG, signalling forward unconditional removed and configured signalling none.

Router log viewer shows Default Label not assigned to OBscript.

VRU-PIM says Transport: Connection to host1(192.168.3.141) attempt failed; error 10061. Host1 defined is IP add of ICM and Port is 38001.

 

FYI... I asked you Trace/debug logs unable to download. I found out the solution. Mozilla replaced IE all working smoothly. I tried dumplog utility but I found it bit complex.

TIA

 

Also, I've question How to correlate/search for call logs in UCCE. Any document / steps guiding for tshoot?

 

 

 

regards,

Ritesh Desai.

 

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

The request never comes to ICM script i guess.

i would request you to increase number of IVR ports in campaign configuration from 2 to 5.

 

under route point for transferring to IVR configuration in campaign, you have defined 2 Dialed Number. 3110,6202.

 

what are those DNs...

have you configured them in ICM, and with what peripheral?

please attache the config of them.

Hi Chintan,

 

IVR Ports are 5 till now. 3110 is the dial no defined for this OB Script. I removed 6202DN which was not used any where.

Cross checked with Production system and corrected few errors.

1. TR Label was assigned wrong RC. Corrected.

2. 7777777777 Label Network type & target not defined. Corrected and defined Type: Network VRU & Target CVP.

 

After this changes restarted all services. Test results are same.

Router log viewer shows Default Label not assigned to OBscript.

 

TIA.

 

 

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

Ritesh,

Here is the catch,

when Dialer dials customer after successful caller detection, dialer must transfer them to CVP/IP IVR for IVR campaign.

you must do this with Translation route to VRU node, in IVR script where dialer sends call.

 

after successful translation route with CVP you can use send to vru for sending call to VXML gateway.

so for MR PG to CVP use translation route and from CVP to VXML Gateway use send yo VRU.

 

please have scripting like above and first see if you are able to send calls from Dialer(MR peripheral) to CVP with translation route successfully?

it must work before anything else.

 

Chintan

 

Hi Chintan,

Thanks for guidance. Honestly, am getting more confused. First of all im getting error "No default label assigned to OB script having DN 3110" which needs to be solved to mode ahead. :( :|

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

no default label means script is abruptly ending without target.

your translation must work for this error to disappear, using default label is not the solution to problem.

 

check the outbound campaign guide and check the sample script you can do for IVR based campaign.

and this script must be associated with the DN you have configured for IVR campaign in ICM.

Hi Chintan,

 

According to OB Guide from Cisco website, I found below missing config.

  1. Type2 Network VRU configured for MR-PIM. PG2A Peripheral (Dialer) Changed from Type10 to Type2.
  2. In CUCM under Device> Device settings, SIP Normalization Script added. Associated with SIP Outbound Trunk.
  3. CPA disabled in ICM Outbound campaign.
  4. AutoAnswerCall registry value changed to 0 from 1 for Dialer.
  5. Blendedagent_addecc.reg.txt file renamed (removed .txt). CTIOS_Instance named added. Executed Blendedagent_addecc.reg file and ECC Variables updated.
  6. Router Registry Key changed.
  7. Route Point added for Tfr to IVR campaign 3110. Route Point added for Tfr to IVR campaign n 3999. DN is not applicable for Tfr to IVR campaign scenario.
  8. Modification of Local Static Route File "DNPHost.txt"

2???,    192.168.3.134,    Transferring Outbound calls to Agent Extensions (2003 Agent Extn configured on CUCM. Agent login on OB CAD using 2003 Extn).
3???,    192.168.3.134,    CTI Route points on CUCM (3999 for Agent NA).
3110,    192.168.3.141,    Transfer to IVR Campaign (3110 defined for After AMD Tfr to IVR campaign).

3.134 is CUCM and 3.141 is ICM.

No Luck.

TIA.

 

 

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

Hi Chintan,

I referred to OB guide for Tfr to IVR campaign. Crossed checked prod system confg and mine.

Debug done on 3 devices (Audio Codes GTW, CUBE and BADialer).

Any hint for below...

AudioCodes Gateway debug:

9d:0h:23m:43s (      lgr_flow)(1906981   )  |       |       #32:Call changing states from:NewCallState_IP2Tel to:InitiatedState_IP2Tel [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906982   )  |       #30:SETUP_EV   : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1906983   )  pstn send --> PlaceCall: Trunk:0 BChannel:1 ConnID:0 SrcPN= SrcSN= DstPN=9920867686 DstSN= SrcNT=0 SrcNP=0 SrcPres=0 SrcScrn=0 DstNT=0 DstNP=0 ServiceCap=M RdrctNum= RdNT=0 RdNP=0 RdPres=0 RdScrn=0 RdRsn=-1 Excl=1 Display= IE= UUIE=0,, CLIRReason:-1 OrigPN= OLI=-1 OffhookInd=0 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906984   )  #30:LOCAL_CALL_PROCEEDING_EV(Trunk:0 Conn:0 Bchannel:1 TpEv=73) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906985   )  |       #30:LOCAL_CALL_PROCEEDING_EV [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906986   )  |       #30:PROGRESS_INDICATOR_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906987   )  |       |       #32:PROGRESS_INDICATOR_EV(PI=8)(PC=-1):(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906988   )  |       #30:PROCEEDING_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906989   )  |       |       #32:PROCEEDING_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906990   )  |       |       |       #32:PROCEEDING_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (     sip_stack)(1906991   )  SIPSDPSession#32 - Changing state from SIP_MEDIA_OFFERED to SIP_MEDIA_COMPLETED [Time: 00:23:43]
9d:0h:23m:43s (   lgr_stk_ses)(1906992   )  DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 00:23:43]
9d:0h:23m:43s (   lgr_stk_ses)(1906993   )  DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_stk_ses)(1906994   )  <SESSION #32> SendToCall - event: DTMF_CONTROL_EV  m_Call = 104137924 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906995   )  |       |       #32:DTMF_CONTROL_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (   lgr_stk_ses)(1906996   )  <SESSION #32> SendToCall - event: OPEN_LOGICAL_CHANNEL_ACK_EV  m_Call = 104137924 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906997   )  |       |       #32:OPEN_LOGICAL_CHANNEL_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906998   )  |       #30:OPEN_LOGICAL_CHANNEL_ACK_EV   : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1906999   )  |       #30:OPEN_VOICE (IP:192.168.1.1, RTP:23940, RTCP:23941, VoiceCoder:g711Ulaw64k20, VbdCoder: InvalidCoder255, Dtmf:gwRFC2833RalayDTMF,Rx payload:101,Tx payload:101 ,RTPmode:1, FaxTransportType: 1,AVoIPMediaType: gwMediaTypeAudioOnly) [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907000   )  #30:ActivateChannel: Socks=15 CID=30 Trunk:0 BChannel:1 RemoteIP=192.168.1.1 RemotePort=23940 RemoteT38IP= RemoteT38Port=0 RemoteRTCPIP=192.168.1.1 RemoteRTCPPort=23941 FaxModemDet=NO_FAX_MODEM_DETECTED [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907001   )  Open channel: IsVoiceOn: 1, IsT38On: 0, IsVbdOn: 0, IsVideoOn: 0 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907002   )  #30:OpenChannel:on Trunk 0 BChannel:1 CID=30 with VoiceCoder: g711Ulaw64k20 VbdCoder: InvalidCoder255 DetectorSide: 0 FaxModemDet NO_FAX_MODEM_DETECTED [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907003   )  #30:OpenChannel VoiceVolume= 1, DTMFVolume = -11, InputGain = 0, RTPRedundancyDepth = 0 FlashHookPeriod = 700 AgcCmd = 0x13180000
 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907004   )  OpenChannel, CoderType = 1, Interval = 3, M = 1
 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907005   )  #30:FAXTransportType = 1 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907006   )  #30:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=1, VxxTranType=2, VoiceVol= 1, DTMFVol=-11, InGain=0, RTPRedDepth=0, ECE=1, SCE=0, ECNlpMode=0, DJBufMinDelay=70, DJBufOptFac=10) [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907007   )  Detectors: Amd:On=0,Diretion=0, Ans:On=0,Direction=0 En:On=0,Direction=0 Board IBScmd:0xa1 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907008   )  Setting ActivateRTP_RTCPCmd.Cmd.IpTosFieldInUdpPacket to 184 [Time: 00:23:43]
9d:0h:23m:43s (   lgr_psbrdif)(1907009   )  #30:ActivateChannel:RtpPayload: 0 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907010   )  |       |(SIPTU#7)ALERT_REQ State:Invited(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907011   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:43]
9d:0h:23m:43s SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK32FDE5
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 101 INVITE
Contact: <sip:1030@192.168.1.233>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 1
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Type: application/sdp
Content-Length: 267

v=0
o=AudiocodesGW 564650208 564634533 IN IP4 192.168.1.233
s=Phone-Call
c=IN IP4 192.168.1.233
t=0 0
m=audio 6300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6301 IN IP4 192.168.1.233
 [Time: 00:23:43]
9d:0h:23m:43s (     sip_stack)(1907013   )  
UdpRtxMngr::Transmit 183 Response 101 INVITE Rtx Left: 6 Dest: 192.168.1.1:5060 CallID: (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907014   )  ---- Incoming SIP Message from 192.168.1.1:56217 to SIPInterface #0 ---- [Time: 00:23:43]
9d:0h:23m:43s PRACK sip:1030@192.168.1.233:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3300
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Date: Fri, 11 Sep 2015 11:20:23 GMT
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 102 PRACK
RAck: 1 101 INVITE
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0

 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907016   )  |       |(SIPTU#7)PRACK State:Invited(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:43s (     sip_stack)(1907017   )  UdpRtxMngr::Remove 183 Response 101 INVITE [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907018   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:43]
9d:0h:23m:43s SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3300
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 102 PRACK
Contact: <sip:1030@192.168.1.233>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Length: 0

 [Time: 00:23:43]
9d:0h:23m:43s (      lgr_flow)(1907020   )  |       |       |       #32:SIP_PRACK_RESPONDED_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:43]
9d:0h:23m:44s (      lgr_flow)(1907021   )  #30:LOCAL_ALERTING_EV(Trunk:0 Conn:0 Bchannel:1 TpEv=69) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907022   )  |       #30:LOCAL_ALERTING_EV [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907023   )  |       #30:PROGRESS_INDICATOR_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907024   )  |       |       #32:PROGRESS_INDICATOR_EV(PI=8)(PC=-1):(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907025   )  |       #30:ALERT_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907026   )  |       |       #32:ALERT_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907027   )  |       |       #32:Call changing states from:InitiatedState_IP2Tel to:AlertingState_IP2Tel [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907028   )  |       |       |       #32:ALERT_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (   lgr_stk_ses)(1907029   )  DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 00:23:44]
9d:0h:23m:44s (   lgr_stk_ses)(1907030   )  DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 00:23:44]
9d:0h:23m:44s (   lgr_stk_ses)(1907031   )  <SESSION #32> SendToCall - event: DTMF_CONTROL_EV  m_Call = 104137924 [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907032   )  |       |       #32:DTMF_CONTROL_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (   lgr_stk_ses)(1907033   )  <SESSION #32> SendToCall - event: OPEN_LOGICAL_CHANNEL_ACK_EV  m_Call = 104137924 [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907034   )  |       |       #32:OPEN_LOGICAL_CHANNEL_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907035   )  |       |(SIPTU#7)ALERT_REQ State:Invited(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907036   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:44]
9d:0h:23m:44s SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK32FDE5
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 101 INVITE
Contact: <sip:1030@192.168.1.233>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 2
x-channel: ds/ds1-0/1;IP=192.168.1.233
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Type: application/sdp
Content-Length: 267

v=0
o=AudiocodesGW 564650208 564634533 IN IP4 192.168.1.233
s=Phone-Call
c=IN IP4 192.168.1.233
t=0 0
m=audio 6300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6301 IN IP4 192.168.1.233
 [Time: 00:23:44
9d:0h:23m:44s (     sip_stack)(1907038   )  
UdpRtxMngr::Transmit 180 Response 101 INVITE Rtx Left: 6 Dest: 192.168.1.1:5060 CallID: (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907039   )  ---- Incoming SIP Message from 192.168.1.1:56217 to SIPInterface #0 ---- [Time: 00:23:44]
9d:0h:23m:44s PRACK sip:1030@192.168.1.233:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK331475
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Date: Fri, 11 Sep 2015 11:20:23 GMT
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 103 PRACK
RAck: 2 101 INVITE
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0

 [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907041   )  |       |(SIPTU#7)PRACK State:Invited(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:44s (     sip_stack)(1907042   )  UdpRtxMngr::Remove 180 Response 101 INVITE [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907043   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:44]
9d:0h:23m:44s SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK331475
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 103 PRACK
Contact: <sip:1030@192.168.1.233>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Length: 0

 [Time: 00:23:44]
9d:0h:23m:44s (      lgr_flow)(1907045   )  |       |       |       #32:SIP_PRACK_RESPONDED_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:44]
9d:0h:23m:53s [PType=9] (      lgr_flow)(1907046   )  ListeningTcpTransportObject#4013- CTcpSocket::HandleSocketEvent socket connected [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907047   )  new TcpTransportObject created - #92 [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907048   )  New Socket buffer allocated - #290 [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907049   )  ---- Incoming SIP Message from 192.168.1.71:5060 to SIPInterface #0 ---- [Time: 00:23:53]
9d:0h:23m:53s OPTIONS sip:Unknown@192.168.1.71 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.71:5060;rport;branch=z9hG4bK8845372295db6d1943efa82ea1d148f2
From: <sip:Unknown@192.168.1.71>;tag=6f3fbedfb58eda80
To: <sip:Unknown@192.168.1.71>
Call-ID: 5a03a2fb1bc0e20fff5b1523b6b49da7@192.168.1.71
CSeq: 1705459707 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

 [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907051   )  |       | new GetNewIndTransaction created - #63 [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907052   )  new AcSIPDialogAPI created - #64 [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907053   )  |       |(SIPTU#63)OPTIONS State:DialogIdle() [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907054   )  SIPDialog(#63) changes state from DialogIdle to DialogInitiated [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907055   )  ServicesMngr::GetEndPoint PhoneNum = Unknown
 [Time: 00:23:53]
9d:0h:23m:53s (   lgr_psbrdif)(1907056   )  GetTrunkGroupId- TrunkGroup:1 found DstNum:Unknown DstPfx:* SrcNum:Unknown SrcPfx:* SrcIp:c0a80147 SrcIpPfx:* [Time: 00:23:53]
9d:0h:23m:53s (   lgr_psbrdif)(1907057   )  Current trunks status: 10 [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907058   )  |       |(SIPTU#63)GENERAL_RESPONSE_REQ State:DialogInitiated(5a03a2fb1bc0e20fff5b1523b6b49da7@192.168.1.71) [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907059   )  SIPDialog(#63) changes state from DialogInitiated to DialogConnected [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907060   )  ---- Outgoing SIP Message to 192.168.1.71:5060 from SIPInterface #0 ---- [Time: 00:23:53]
9d:0h:23m:53s SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.71:5060;rport;branch=z9hG4bK8845372295db6d1943efa82ea1d148f2
From: <sip:Unknown@192.168.1.71>;tag=6f3fbedfb58eda80
To: <sip:Unknown@192.168.1.71>;tag=1c724193317
Call-ID: 5a03a2fb1bc0e20fff5b1523b6b49da7@192.168.1.71
CSeq: 1705459707 OPTIONS
Contact: <sip:192.168.1.233;transport=tcp>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
X-Resources: telchs=29/1;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 291

v=0
o=AudiocodesGW 724217956 724217592 IN IP4 192.168.1.233
s=Phone-Call
c=IN IP4 192.168.1.233
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtp
9d:0h:23m:53s (      lgr_flow)(1907062   )  |       |(SIPTU#63)DIALOG_DISCONNECT_REQ State:DialogConnected(5a03a2fb1bc0e20fff5b1523b6b49da7@192.168.1.71) [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907063   )  SIPDialog(#63) changes state from DialogConnected to DialogDisconnected [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907064   )  AcSIPStackAPI::FreeDialogAPI - #64 [Time: 00:23:53]
9d:0h:23m:53s (      lgr_flow)(1907065   )  |       | TransactionUserMngr::ReturnDialog - #63 [Time: 00:23:53]
9d:0h:23m:53s (     sip_stack)(1907066   )  SIPDialog(#63) changes state from DialogDisconnected to DialogIdle [Time: 00:23:53]
9d:0h:23m:55s (      lgr_flow)(1907067   )  #30:LOCAL_CALL_CONNECTED_EV(Trunk:0 Conn:0 Bchannel:1 TpEv=65) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907068   )  |       #30:LOCAL_CALL_CONNECTED_EV [Time: 00:23:55]
9d:0h:23m:55s (   lgr_psbrdif)(1907069   )  #30:PSOSBoardInterface::StopPlayTone- Called [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907070   )  |       #30:CONNECT_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (   lgr_psbrdif)(1907071   )  Restart ECE control (CID=30) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907072   )  |       |       #32:CONNECT_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907073   )  |       |       #32:Call changing states from:AlertingState_IP2Tel to:ConnectedState [Time: 00:23:55]
9d:0h:23m:55s |CALL_CONNECT|30   |0              |0    |1    |0    |1    |ISDN      |IP   |192.168.1.1         |192.168.1.233       |0    |0    |                    |                    |0    |0    |9920867686          |9920867686          |0    |g711Ulaw64k    |20   |192.168.1.1         |23940|UNKN |REASON N/A                              |0    |0         |600       |0         |-1            |ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1     |                                    |                                    |                                    |0        |0        |0         |0             |-1             |0    |0    |                    |-1                  
 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907075   )  |       |       |       #32:CONNECT_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907076   )  |       |(SIPTU#7)CONNECT_REQ State:Invited(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907077   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:55]
9d:0h:23m:55s SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK32FDE5
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 101 INVITE
Contact: <sip:1030@192.168.1.233>
Supported: em,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
x-channel: ds/ds1-0/1;IP=192.168.1.233
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Type: application/sdp
Content-Length: 267

v=0
o=AudiocodesGW 564650208 564634533 IN IP4 192.168.1.233
s=Phone-Call
c=IN IP4 192.168.1.233
t=0 0
m=audio 6300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6301 IN IP4 192
9d:0h:23m:55s (     sip_stack)(1907079   )  
UdpRtxMngr::Transmit 200 Response 101 INVITE Rtx Left: 6 Dest: 192.168.1.1:5060 CallID: (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (     sip_stack)(1907080   )  SIPCall(#7) changes state from Invited to LocalAccepted [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907081   )  DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_RFC2833_SUPPORTED [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907082   )  DtmfCapNegotiationAlgorithm :: TxRtpRfc2833Payload = 101 [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907083   )  <SESSION #32> SendToCall - event: DTMF_CONTROL_EV  m_Call = 104137924 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907084   )  |       |       #32:DTMF_CONTROL_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907085   )  <SESSION #32> SendToCall - event: OPEN_LOGICAL_CHANNEL_ACK_EV  m_Call = 104137924 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907086   )  |       |       #32:OPEN_LOGICAL_CHANNEL_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907087   )  ---- Incoming SIP Message from 192.168.1.1:56217 to SIPInterface #0 ---- [Time: 00:23:55]
9d:0h:23m:55s ACK sip:1030@192.168.1.233:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3321A4A
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Date: Fri, 11 Sep 2015 11:20:23 GMT
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

 [Time: 00:23:55]
9d:0h:23m:55s (     sip_stack)(1907089   )  UdpRtxMngr::Remove 200 Response 101 INVITE [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907090   )  |       |(SIPTU#7)ACK State:LocalAccepted(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (     sip_stack)(1907091   )  SIPCall(#7) changes state from LocalAccepted to Connected [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907092   )  |       |       |       #32:SIP_ACK_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907093   )  <SESSION #32> SendToCall - event: CONNECT_ACK_EV  m_Call = 104137924 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907094   )  |       |       #32:CONNECT_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907095   )  |       #30:CONNECT_ACK_EV   : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907096   )  ---- Incoming SIP Message from 192.168.1.1:56217 to SIPInterface #0 ---- [Time: 00:23:55]
9d:0h:23m:55s BYE sip:1030@192.168.1.233:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3361966
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Date: Fri, 11 Sep 2015 11:20:23 GMT
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1441970435
CSeq: 104 BYE
Reason: Q.850;cause=16
Content-Length: 0

 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907098   )  |       |(SIPTU#7)BYE State:Connected(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (     sip_stack)(1907099   )  SIPCall(#7) changes state from Connected to Disconnected [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907100   )  |       |       |       #32:SIP_DISCONNECT_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (   lgr_stk_ses)(1907101   )  <SESSION #32> SendToCall - event: RELEASE_EV  m_Call = 104137924 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907102   )  |       |       #32:RELEASE_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907103   )  |       |       #32:Call changing states from:ConnectedState to:DisconnectingState [Time: 00:23:55]
9d:0h:23m:55s |CALL_END    |30   |0              |0    |1    |0    |1    |ISDN      |IP   |192.168.1.1         |192.168.1.233       |0    |0    |                    |                    |0    |0    |9920867686          |9920867686          |0    |g711Ulaw64k    |20   |192.168.1.1         |23940|IP   |GWAPP_NORMAL_CALL_CLEAR                 |0    |0         |604       |0         |-1            |ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1     |                                    |                                    |                                    |0        |0        |0         |0             |-1             |0    |0    |                    |0                   
 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907105   )  |       |       #32:RELEASE_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907106   )  |       #30:RELEASE_EV  GWAPP_NORMAL_CALL_CLEAR : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907107   )  |       #30:Close voice Channel [Time: 00:23:55]
9d:0h:23m:55s (   lgr_psbrdif)(1907108   )  #30:StopRTP_RTCP on channel 30 [Time: 00:23:55]
9d:0h:23m:55s (   lgr_psbrdif)(1907109   )  pstn send --> PSTNDisconnectCall() Trunk:0 Conn:0 BChannel:1 Reason:16 Loc:-1 Des:-1 rc:0 [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907110   )  |       #30:RELEASE_ACK_EV (send)  : (ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907111   )  |       |       #32:RELEASE_ACK_EV:(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907112   )  |       |       |       #32:RELEASE_ACK_EV(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907113   )  |       |(SIPTU#7)DISCONNECT_RESPONSE State:Disconnected(ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1) [Time: 00:23:55]
9d:0h:23m:55s (      lgr_flow)(1907114   )  ---- Outgoing SIP Message to 192.168.1.1:5060 from SIPInterface #0 ---- [Time: 00:23:55]
9d:0h:23m:55s SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3361966
From: <sip:192.168.1.1>;tag=6C3E2864-11FF
To: <sip:9920867686@192.168.1.233>;tag=1c564549426
Call-ID: ED97FECD-57AD11E5-B6ACEAF4-163618@192.168.1.1
CSeq: 104 BYE
Contact: <sip:1030@192.168.1.233>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.018.005
Content-Length: 0

CUBE debug:

OPTIONS sip:192.168.1.1 SIP/2.0Via: SIP/2.0/UDP 192.168.3.141:58800;branch=z9hG4bK-d8754z-a764e8374d6b3746-1---d8754z-;rportMax-Forwards: 70Contact: <sip:sipping@192.168.3.141:58800>To: <sip:192.168.1.1>From: <sip:sipping@192.168.3.141>;tag=7d320069Call-ID: 0b47b175-a4419500-a253a474-5c7af855CSeq: 1 OPTIONSAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONSSupported: timer, resource-priority, replacesUser-Agent: Cisco-SIPDialer/UCCE10.0Content-Length: 0
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x2A062DD8) with key=[193111] to table
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.3.141,Port 58800, Transport 1, SentBy Port 58800
Sep 11 16:50:24.254: //-1/EE3C2E0EB6AD/SIP/State/sipSPIChangeState: 0x2A062DD8 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.3.141,Port 58800, Transport 1, SentBy Port 5060
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.3.141,Port 58800, Transport 1, SentBy Port 58800
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/ccsipInitPldCallInfo: non-numeric calling number: sipping
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : sipping
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetFromCalledPartyId: P-Called-Party-ID header not found
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetPeerByCalledPartyId: P-Called-Party-ID not found or parse error
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
Sep 11 16:50:24.
CUBE2921#254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 255
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Error/rtpAvpCodec_to_voipCodec: Unexpected RTP PayloadType :255 in SDP Body
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.1
Sep 11 16:50:24.254: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_OPTIONS_RESP
Sep 11 16:50:24.254: //193401/EE3C2E0EB6AD/SIP/Info/sact_idle_new_message_options: ccsip_api_options_ind returned: SIP_SUCCESS

 

====================================================

BADialer debug:

Dialog ID is generating but throwing this error: "ClientNonInviteTrying unreliable target=[ V4 0.0.0.0:0 UNKNOWN_TRANSPORT target domain=unspecified"

 

I suspect Gateway is not doing CPA functionality. Customer goes off hook on mobile phone immediately call disconnects.

On AudioCodes am seeing Dialog disconnected.

Appreciate your support. Thanks.

 

 

regards,

Ritesh Desai

 

 

 

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

Chintan,

CPA and IP AMD(Tfr to IVR RP) now enabled in Campaign. Now call is not disconnecting but Human voice is not getting detected.

 

Do I need to configure CPA function in CUBE?

CUBE version (C2900-UNIVERSALK9-M), Version 15.0(1)M1. I dont see call-progress analysis command in Router.

TIA

 

 

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

please disable CPA and AMD detection on campaign. they are not required unless you have specific requirement.

if after disabling also it doesn't work then we have to look at where in script its failing.

 

i am kind of away from computer till sunday but will look into the thread when possible.

Chintan,

Appreciate your feedback friend. Found this place where you T-shoot similar problem like me.

https://supportforums.cisco.com/discussion/12211111/transfer-ivr-outbound-campaign-does-not-work

I had confusion and cleared.

1. dialer initiates call to customer,and no agents are available.

2. dialer initiates transfer to IVR campaign, initiates route request on configured DN(this will be for MR PG) in Skill Group selection tab in campaign configuration.
==> No defined in campaign config (After AMD & Tfr to IVR campaign as 3110). 3110 wildcard entry inserted in DNPHost file as

"3110,192.168.3.141,To MRPG"

3. DN on MR PG runs scheduled script for the call type.
==> 3110 is the DN for ICM Script.

4. Script executes Translation route to VRU node.

translation route configuration is complex, look at the screen shot attached.
it is configured for destination Trunk Group(Group of CVP peripheral) where your call is supposed to land.

look at the label node:

this is the label will be passed back to you routing client for temporary destination, in your case it should go to MR PG.
look at Peripheral target, it is the DNIS expected  when translation routed call arrives on destined peripheral and and asks

instruction to ICM.

then ICM will be able to match the call and resume its routing script.
)


5. TR label is returned back to MR PG. and MR PG forwards it to dialer.
==> (77001 - 77005) is my TR DNIS and Label.

6. Dialer looks at its DNP host file, and finds the gateway where this label is supposed to go. (this will be SIP refer which

dialer will initiate)
==> 7700?,    192.168.1.1,    To CUBE for OB Transfer to IVR campaign.  In my DNPHost file entry.

7.gateway matches its dial-peer for the label and sends a call to configured destination target. in your case it will be CVP

configured as a peripheral target.
==>
dial-peer voice 777777 voip
 description *****From Dialer to CVP for OUTBOUND_IVRcampaign*****
 destination-pattern 7700.
 session protocol sipv2
 session target ipv4:192.168.3.142
 session transport udp
 voice-class sip rel1xx require "100rel"
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad


8. Under CVP call server configuration, under ICM tab, this DNIS has to configured as TR routed DNIS. look at attached.
==> Range 77001 - 77005 DNIS range added.

 

DNPHost file entry looks like

3110,    192.168.3.141,    To MRPG for OB Transfer to IVR campaign.
7700?,    192.168.1.1,    To CUBE for OB Transfer to IVR campaign.

 

 

TIA,

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

you are on the track Ritesh,

you dont have to put the 3110 in DNP host file as that will be automatically detected by dailer for requesting instruction for either agent or IVR port.

the DNP host file is only needed for SIP refer which dialer sends out when it gets SIP Invite.

has your dialer started working?

 

 

Thanks for fast response Chintan.

I'll remove 3110 DN for Tfr to IVR campaign. Unable to check Dialer, PRI is down. Will update you soon.

 

 

regards,

Ritesh Desai

 


 

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

Hi Chintan,

Am back. I was on PL for 10days.

Still above error is unsolved. : No default label available for DN 3110 (ID5006). Any guide to this.

 

TIA.

 

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

what is the status of your translation route? is it working, does call come out of success node of translation route to VRU node?

 

Chintan