No Redundancy for Inbound Calls

Dear all,
We have 2 CallManagers at our organization. Incoming calls are routed to Attendant Console to listening Music on Hold. Whenever the Publisher CallManager is down, calls coming into our organization receive a busy signal. I thought we were setup for redundancy, but I found out the hard way that this was not the case.
We have a 2911 router which routes incoming phone traffic into our organization. On it are two dial peers which route our organization?s incoming calls to the CallManagers. The primary dial peer is supposed to route to the publishe and the secondary dial peer to the subscriber. Below are the 2 dial peers which route our traffic.
voice class h323 1
h225 timeout tcp establish 3
dial-peer voice 11 voip
preference 2
destination-pattern 3.......
session target ipv4:10.1.11.30 ( publiser)
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 10 voip
preference 1
destination-pattern 3.......
session target ipv4:10.1.11.31 ( subcriber)
dtmf-relay h245-alphanumeric
codec g711alaw
If i delete dial peer voice 11, incoming call can route to subcriber successfully
I would appreciate anyone taking the time to look at them and offer any suggestions.
CCM 10.0
I got debug for this as below:
TSI-DC-VGR01-0#end
Translating "end"
% Bad IP address or host name
Translating "end"
% Unknown command or computer name, or unable to find computer address
TSI-DC-VGR01-0#
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=0905916072
   cisco-anitype=2
   cisco-aniplan=1
   cisco-anipi=0
   cisco-anisi=3
   dest=38236688
   cisco-desttype=0
   cisco-destplan=1
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/cc_api_call_setup_ind_common:   Interface=0x2BA17414, Call Info(   Calling Number=0905916072,(Calling Name=)(TON=National, NPI=ISDN, Screenin
etwork, Presentation=Allowed),
   Called Number=38236688(TON=Unknown, NPI=ISDN),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinatioag=TRUE,
   Incoming Dial-peer=9, Progress Indication=NULL(0), Calling IE Present=TRUE
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=F
E), Call Id=-1
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/ccCheckClipClir:
   In: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, P
entation=Allowed)
Jul 30 07:48:09.070: //-1/AEEEB6D6813B/CCAPI/ccCheckClipClir:
   Out: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network,
sentation=Allowed)
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :cc_get_feature_vsa malloc success
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: cc_get_feature_vsa count is 1
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :FEATURE_VSA attributes are: feature_name:0,feature_time
0612504,feature_id:1877
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Netw
, Presentation=Allowed),
   Called Number=38236688(TON=Unknown, NPI=ISDN))
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_process_call_setup_ind:
   Event=0x2B9DF948
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 38236688
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetContext:
   Context=0x300C4B5C
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 1877 with tag 9 to app "_ManagedAppProcess_Default"
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=10, Params=0x300C606C, Progress Indication=NULL(0)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCheckClipClir:
   In: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, P
entation=Allowed)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCheckClipClir:
   Out: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network,
sentation=Allowed)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
   Destination Pattern=3......., Called Number=38236688, Digit Strip=FALSE
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
   Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, Prese
tion=Allowed),
   Called Number=38236688(TON=Unknown, NPI=ISDN),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=AEEEB6D6-16F4-11E4-813B-A4934C055B80, Outgoing Dial-peer=10
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=0905916072
   cisco-anitype=2
   cisco-aniplan=1
   cisco-anipi=0
   cisco-anisi=3
   dest=38236688
   cisco-desttype=0
   cisco-destplan=1
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x312C8048, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=0905916072,(Calling Name=)(TON=National, NPI=IS
Screening=Network, Presentation=Allowed),
   Called Number=38236688(TON=Unknown, NPI=ISDN), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-
r=10, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, A
ication Call Id=)
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :cc_get_feature_vsa malloc success
Jul 30 07:48:09.074: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.074: cc_get_feature_vsa count is 2
Jul 30 07:48:09.074: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.074: :FEATURE_VSA attributes are: feature_name:0,feature_time
0612952,feature_id:1878
Jul 30 07:48:09.074: //1878/AEEEB6D6813B/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Jul 30 07:48:09.074: //1878/AEEEB6D6813B/CCAPI/ccCallSetContext:
   Context=0x300C601C
Jul 30 07:48:09.074: //1877/AEEEB6D6813B/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=10
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnected:
   Cause Value=102, Interface=0x2BA17414, Call Id=1877
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=FALSE, Cause Value=102, Retry Count=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/ccCallDisconnect:
   Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconne
Cause=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/ccCallDisconnect:
   Cause Value=102, Call Entry(Responsed=FALSE, Cause Value=102)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/ccCallDisconnect:
   Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconne
Cause=102)
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/ccCallDisconnect:
   Cause Value=102, Call Entry(Responsed=TRUE, Cause Value=102)
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x312C8048, Tag=0x0, Call Id=1878,
   Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Jul 30 07:48:19.386: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.386: :cc_free_feature_vsa freeing 2C24DB50
Jul 30 07:48:19.386: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.386: vsacount in free is 1
Jul 30 07:48:19.394: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x2BA17414, Tag=0x0, Call Id=1877,
   Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
Jul 30 07:48:19.398: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Jul 30 07:48:19.398: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.398: :cc_free_feature_vsa freeing 2C24D990
Jul 30 07:48:19.398: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Dear Okanlawon & islam.kamal,
Both of you are correct. I used your command and it worked now. It also help me solved the problem related to Music On Hold cause i use g711ulaw ( MoH wont work with incoming call).
I used c2900-universalk9-mz.SPA.151-4.M4.bin
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M4,
EASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2012 by Cisco Systems, Inc.
Compiled Tue 20-Mar-12 18:57 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M15, RELEASE SOFTWARE (fc1)
Thank you very much ! You are the god !
ThanhNT

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GENERIC:
SetupTime=12:24:35.980 AEDT Thu Jan 30 2014
Index=1
PeerAddress=0204002244
PeerSubAddress=
PeerId=31
PeerIfIndex=192
LogicalIfIndex=0
ConnectTime=12:24:36.080 AEDT Thu Jan 30 2014
CallDuration=00:00:43 sec
CallState=4
CallOrigin=2
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InfoType=speech
TransmitPackets=2196
TransmitBytes=351360
ReceivePackets=2194
ReceiveBytes=351040
VOIP:
ConnectionId[0x1D4C7D6F 0x888411E3 0xA9FBFF95 0xBEE5C848]
IncomingConnectionId[0x1D4C7D6F 0x888411E3 0xA9FBFF95 0xBEE5C848]
CallID=5201
GlobalCallId=[0x1D4C7D6F 0x888411E3 0xA9FEFF95 0xBEE5C848]
CallReferenceId=0
CallServiceType=Unknown
RTP Loopback Call=FALSE
RemoteIPAddress=10.193.202.100
RemoteUDPPort=14804
RemoteSignallingIPAddress=10.193.202.100
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.193.202.100
RemoteMediaPort=14804
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=inband-voice
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
[email protected]
SessionTarget=10.193.202.100
SafEnabled=FALSE
OnTimeRvPlayout=0
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=0 ms
LoWaterPlayoutDelay=0 ms
TxPakNumber=0
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=0
TxVoiceDuration=0
RxPakNumber=0
RxSignalPak=0
RxComfortNoisePak=0
RxDuration=0
RxVoiceDuration=0
RxOutOfSeq=0
RxLatePak=0
RxEarlyPak=0
RxBadProtocol=0
PlayDelayCurrent=0
PlayDelayMin=0
PlayDelayMax=0
PlayDelayClockOffset=0
PlayDelayJitter=0
PlayErrPredictive=0
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=0
PlayErrRetroactive=0
PlayErrTalkspurt=0
ClockAdjustDuration=0
EarlyPacketDuration=0
LateDiscardDuration=0
SilenceDiscardDuration=0
JBuffDecreaseDuration=0
JBuffIncreaseDuration=0
OutSignalLevel=0
InSignalLevel=0
LevelTxPowerMean=0
LevelRxPowerMean=0
LevelBgNoise=0
ERLLevel=0
ACOMLevel=0
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=0 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
SRTP = off
TextRelay = off
VAD = disabled
CoderTypeRate=g711alaw
CodecBytes=160
Media Setting=flow-through
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=0204002244
OriginalCallingOctet=0x0
OriginalCalledNumber=0498870440
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x80
TranslatedCallingNumber=0204002244
TranslatedCallingOctet=0x0
TranslatedCalledNumber=0498870440
TranslatedCalledOctet=0x0
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0x80
GwReceivedCalledNumber=0498870440
GwReceivedCalledOctet3=0x0
GwReceivedCallingNumber=0204002244
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x80
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurCallTimestamp=
LongDurcallDuration=
Username=0204002244
MlppServiceDomainNW=0 (none)
MlppServiceDomainID=
PrecedenceLevel=0 (PRECEDENCE_LEVEL_NONE)
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SetupTime=12:24:35.994 AEDT Thu Jan 30 2014
Index=1
PeerAddress=2440
PeerSubAddress=
PeerId=400
PeerIfIndex=202
LogicalIfIndex=0
ConnectTime=12:24:36.074 AEDT Thu Jan 30 2014
CallDuration=00:00:43 sec
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=2196
TransmitBytes=351360
ReceivePackets=4396
ReceiveBytes=703360
VOIP:
ConnectionId[0x1D4C7D6F 0x888411E3 0xA9FBFF95 0xBEE5C848]
IncomingConnectionId[0x1D4C7D6F 0x888411E3 0xA9FBFF95 0xBEE5C848]
CallID=5202
GlobalCallId=[0x1D4C7D6F 0x888411E3 0xA9FEFF95 0xBEE5C848]
CallReferenceId=0
CallServiceType=Unknown
RTP Loopback Call=FALSE
RemoteIPAddress=10.66.22.20
RemoteUDPPort=31334
RemoteSignallingIPAddress=10.66.22.20
RemoteSignallingPort=5060
RemoteMediaIPAddress=10.66.21.20
RemoteMediaPort=31334
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE
AnnexE=FALSE
Separate H245 Connection=FALSE
H245 Tunneling=FALSE
SessionProtocol=sipv2
[email protected]
SessionTarget=10.66.22.20
SafEnabled=FALSE
OnTimeRvPlayout=0
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=0 ms
LoWaterPlayoutDelay=0 ms
TxPakNumber=0
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=0
TxVoiceDuration=0
RxPakNumber=0
RxSignalPak=0
RxComfortNoisePak=0
RxDuration=0
RxVoiceDuration=0
RxOutOfSeq=0
RxLatePak=0
RxEarlyPak=0
RxBadProtocol=0
PlayDelayCurrent=0
PlayDelayMin=0
PlayDelayMax=0
PlayDelayClockOffset=0
PlayDelayJitter=0
PlayErrPredictive=0
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=0
PlayErrRetroactive=0
PlayErrTalkspurt=0
ClockAdjustDuration=0
EarlyPacketDuration=0
LateDiscardDuration=0
SilenceDiscardDuration=0
JBuffDecreaseDuration=0
JBuffIncreaseDuration=0
OutSignalLevel=0
InSignalLevel=0
LevelTxPowerMean=0
LevelRxPowerMean=0
LevelBgNoise=0
ERLLevel=0
ACOMLevel=0
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=0 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
SRTP = off
TextRelay = off
VAD = disabled
CoderTypeRate=g711alaw
CodecBytes=160
Media Setting=flow-through
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CallerName=
CallerIDBlocked=False
OriginalCallingNumber=0204002244
OriginalCallingOctet=0x0
OriginalCalledNumber=0498870440
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x80
TranslatedCallingNumber=00204002244
TranslatedCallingOctet=0x0
TranslatedCalledNumber=2440
TranslatedCalledOctet=0x0
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0x80
GwReceivedCalledNumber=0498870440
GwReceivedCalledOctet3=0x0
GwOutpulsedCalledNumber=2440
GwOutpulsedCalledOctet3=0x0
GwReceivedCallingNumber=0204002244
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x80
GwOutpulsedCallingNumber=00204002244
GwOutpulsedCallingOctet3=0x0
GwOutpulsedCallingOctet3a=0x80
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurCallTimestamp=
LongDurcallDuration=
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MlppServiceDomainNW=0 (none)
MlppServiceDomainID=
PrecedenceLevel=0 (PRECEDENCE_LEVEL_NONE)

No ringbacktone for inbound calls with cucm 8.6

Hi,
we have this problem from many days...
we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
our setup is like below
HO and BR2 having SIP lines and BR1 has PSTN Lines.
we implement greetings for head office and 2 branches at Headoffice Unity.
when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
i tried to change the " Send h225 User Information Message" in service parameters from "Use ANN for Ring Back" to H225 Info for call Progress Tone"
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please can anyone help............................

Hi Carlo,
Thankyou for the Response...
here is the Runn config for BR1 Connected to PSTN lines....
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice class h323 1
h225 timeout tcp establish 3
interface Tunnel100
description " Tunnel JED-RYD "
bandwidth 2048
ip address 10.10.0.1 255.255.255.252
tunnel source 172.31.217.202
tunnel destination 172.31.3.18
interface FastEthernet0/0
description DAMMAM Local LAN
no ip address
duplex auto
speed auto
interface FastEthernet0/0.20
description JEDDAH Local LAN
encapsulation dot1Q 20
ip address 192.168.20.5 255.255.255.0
interface FastEthernet0/0.21
description JEDDAH VOICE VLAN
encapsulation dot1Q 21
ip address 192.168.21.5 255.255.255.0
h323-gateway voip interface
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interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
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network 10.10.0.0 0.0.0.3
network 192.168.20.0
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no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
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no auto-summary
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control-plane
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/2/0
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
connection plar 2022
shutdown
impedance complex2
description STC
voice-port 0/2/1
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
shutdown
impedance complex2
description STC
voice-port 0/3/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
sccp local FastEthernet0/0.21
sccp ccm 192.168.12.190 identifier 1 priority 1 version 5.0.1
sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CONFJEDRAW
associate profile 2 register TRNJED
associate profile 3 register MTPJED
switchover method immediate
switchback method immediate
switchback interval 15
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
shutdown
dspfarm profile 3 mtp
codec g729r8
maximum sessions software 250
associate application SCCP
shutdown
dial-peer voice 1 pots
dial-peer voice 1000 voip
description To CallManager - SBWPMPUB
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 9001 pots
description ** 02-6140294(outgoing) **
destination-pattern [^2].T
port 0/0/1
dial-peer voice 9002 pots
description ** 02-6140295(outgoing) **
destination-pattern [^2].T
port 0/0/2
dial-peer voice 9003 pots
description ** 02-6140296(outgoing) **
destination-pattern [^2].T
port 0/0/3
dial-peer voice 9004 pots
description ** 02-6140293(outgoing) **
destination-pattern [^2].T
port 0/0/0
dial-peer voice 290 pots
incoming called-number .
direct-inward-dial
dial-peer voice 9006 pots
description ** 02-6529323(local) **
destination-pattern [^0].T
port 0/3/0
dial-peer voice 9010 pots
description ** 02-6578249(local) **
destination-pattern [^0].T
port 0/3/1
dial-peer voice 9011 pots
description "to pstn service"
shutdown
destination-pattern 0.T
port 0/3/3
dial-peer voice 9009 pots
description "to pstn service"
shutdown
destination-pattern [^0].T
port 0/3/2
dial-peer voice 9005 pots
destination-pattern .T
dial-peer voice 1001 voip
description To CallManager - Subscriber
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1002 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1003 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad

Lync Dispalying wrong Name for inbound calls

We have Active Directory running on Windows 2012 R2 Domain Controllers. We have Exchange 2010 and LYNC 2013. We have presence integration with our Avaya CM ver 6.x.
We have had a few employee's leave and have re-assigned their desk phone extensions to new employee's. The old employee's were removed from Active Directory and the LYNC server.
On the LYNC 2013 client when a call comes in it is showing the old employee's name on the LYNC Client.
I have deleted the local LYNC Addressbooks at the client.
Still happening.
Where to look for the file containing the old employee that is causing this mismatch.
Thank you,

Hi WillieWinslow,
Agree with Paul.
Before starting a new test, you can check the following items.
1. Make sure the contact who has left the company is not existing in the user’s Outlook Contact folders.
2. (Optional)Run Update-CsUserDataBase to force Back-End to
re-read all the user-related information stored in Active Directory Domain Services.
Delete the .slabs file from the file share sitting on X:\share\1-WebServices-1\ABfiles\000000000\000000000 folder.
3. Run Update-CsAddressBook
4. Delete the test user’s Lync profile on the test machine.
After doing the above, it should be able to solve your problem.
Best regards,
Eric

I'm unable to accept inbound calls on my iPhone 5 for a holden commodore VE - 2012 model - any ideas on how to resolve?

How do I resolve the issue for answering inbound calls on my iPhone 5 when connected to Bluetooth in a holden commodore VE - 2012 model. Outbound calls connect ok.

I'm having the same problem with a 2013 GMC terrain. I don't know if the problem is the car or the iPhone 5. I have been seeing this problem more and more on threads. Can anyone help us with this? Is there any info online from Apple re: new updates that would solve this problem?

Goods Receipt for Inbound Delivery through MIGO

Hi All,
Can anyone pl. explain me in steps the process of GR for Inbound Delivery from MIGO transaction.
Also I need the process of reversing this goods receipt done.
For certain inbound deliveries a dump occurs in the program SAPLIE01 in 'EQUIPMENT_ADAPT_BUSVIEWS' fn module.
I need to analyse the reason for the dump. Here is the extract from the dump
<b>IF WA_EQUI IS INITIAL
keine Equidaten vorhanden => EXCEPTION
---> RAISE ERR_ADAPT_BUSVIEWS . ENDIF. </b>
Advance Thanks

Hi Sanjay and Srikanth,
Thank you for the correction. I was just wondering over this number.
Sanjay, regd the release level, it is 0017 for SAP_APPL.
In brief, when GR is done for an inbound delivery, a fn moudle equipment_adapt_busviews is called, in which if the equipment data of the material (wa_equi) is initial, it gives dump.
1. This fn module is called 3 or 4 times while the posting is done through MIGO
2. In the 4th call, the parameter wa_equi is initial - which gives the dump.
3. While debugging, when I reach this point,if I restart the debugging, I find that the posting is completed !!!! (so something happening after posting of GR gives the dump)
4. I have to reverse this GR for replicating the same again.
Any further clues !!
Regards
Aadarsh

Debugging inbound call - need debug command

I am debugging what I believe to be a dial plan issue on PGW but want to run a debug command on the ITP to show the actual called number of the inbound call.
What is the best debug command to use to see the inbound called and calling numbers of a call.

You are better off capturing an mdl trace from the PGW using per call tracing or using snoop and opening the call in Wireshark.
The ITP will only dump hex values and you will need to decode them yourself since it is not concerned with ISUP layer. You can use the cs7 paklog feature to attach an access list to a linkset and then send the raw data to a syslog server for decoding.
For example:
cs7 paklog x.x.x.x dest-port xxxx src-port yyyy
access-list 2700 instance x permit si all
debug cs7 mtp3 paklog 2700

Use WS_DELIVERY_UPDATE_2 for Inbound Delivery HU Update?

Hello all,
I have used WS_DELIVERY_UPDATE_2 to successfully update Handling units to Outbound delivery and also perform batch split.
Can the same FM be used to update Inbound delivery with Handling units. I will be uploading the HU's through excel file.
Please advice.
Thanks,
Subba

Hi,
Below is code for Inbound Delivery With HU, it worked for me.
gs_head-DELIV_DATE = sy-datum.
gs_head-DELIV_TIME = sy-uzeit.
gs_item-DELIV_ITEM = p_item1.
gs_item-MATERIAL = p_matnr1.
gs_item-DELIV_QTY = p_quan.
gs_item-UNIT = 'EA'.
gs_item-PO_NUMBER = p_po.
gs_item-PO_ITEM = 10.
APPEND gs_item TO gt_item.
CLEAR:gs_item.
CALL FUNCTION 'BBP_INB_DELIVERY_CREATE'
EXPORTING
    IS_INB_DELIVERY_HEADER = gs_head
IMPORTING
    EF_DELIVERY            = gv_vbeln
TABLES
    IT_INB_DELIVERY_DETAIL = gt_item
    RETURN                 = gt_ret1.
gs_refobj-OBJECT = 03.
gs_refobj-OBJKEY = gv_vbeln.
gs_humain-vhilm = p_matnr2.
GS_HUMAIN-HU_STATUS_INIT = 'A'.
gs_humain-STATUS = '0010'.
gs_itempr1-VELIN = 1.
gs_itempr1-QUANTITY = p_quan.
gs_itempr1-meins = 'EA'.
gs_itempr1-matnr = p_matnr1.
gs_itempr1-werks = p_plant.
CALL FUNCTION 'PROCESS_HU_INBOUND_DLVRY'
EXPORTING
    I_HU_ID               = gv_huid (Blank)
    I_NEW_HU              = 'X'
    I_UPDATE_DB           = 'X'
    I_CREATE_HU           = 'X'
    I_REFERENCE_OBJECT    = gs_refobj
    I_HU_HEADER_MAIN      = gs_humain
    I_SAVE_STEP           = 'ITEM'
    I_PACK_ACTIVITY       = 'MAT'
    I_DELIVERY_TYPE       = '7'
    I_PACKING_ACTIVITY    = 'PACK'
    I_PACK_TYPE           = 'MAT'
    IS_ITEM_PROPOSAL      = gs_itempr1
IMPORTING
    O_HU_REF_ID           = gs_erefid
    O_HU_HEADER           = gt_eheader
    O_HU_ITEMS            = gt_eitem
    O_HU_MESSAGES         = gt_emsg[]
TABLES
    ET_CREATED_HUS        = gt_ehus[]
EXCEPTIONS
    INPUT_MISSING         = 1
    NOT_POSSIBLE          = 2
    HEADER_ERROR          = 3
    ITEM_ERROR            = 4
    SERIAL_NR_ERROR       = 5
    FATAL_ERROR           = 6
    OPERATION_FAILED      = 7
    DELIVERY_UPDATE_ERROR = 8
    OTHERS                = 9.
IF SY-SUBRC EQ 0.
CLEAR:gs_vekp.
SELECT SINGLE * FROM vekp INTO gs_vekp WHERE VPOBJKEY = GV_VBELN.
ls_vbkok_wa-vbeln_vl = gv_vbeln.
ls_vbkok_wa-vbeln = gv_vbeln.
ls_vbkok_wa-VBTYP_VL = 7.
ls_vbkok_wa-wabuc = 'X'.
ls_vbkok_wa-spe_auto_gr = 'X'.
ls_vbkok_wa-kzebu = 'X'.
lt_verko-exidv = gs_vekp-exidv.
lt_verko-exida = 'E'.
lt_verko-magrv = 'YB01'.
lt_verko-werks = p_plant.
lt_verko-STATUS = '0010'.
APPEND lt_verko.
lt_verpo-venum = gs_vekp-venum.
lt_verpo-exidv_ob = gs_vekp-exidv.
lt_verpo-exidv = gs_vekp-exidv.
lt_verpo-velin = '1'.
lt_verpo-vbeln = gv_vbeln.
lt_verpo-POSNR = p_item1 .
lt_verpo-TMENG = p_quan.
lt_verpo-VRKME = 'EA'.
lt_verpo-MATNR = p_matnr1.
lt_verpo-werks = p_plant.
APPEND lt_verpo.
lt_vbpok-vbeln_vl = gv_vbeln.
lt_vbpok-posnr_vl = p_item1.
lt_vbpok-vbeln = gv_vbeln.
lt_vbpok-posnn = p_item1.
lt_vbpok-pikmg = p_quan.
lt_vbpok-matnr = p_matnr1.
lt_vbpok-werks = p_plant.
lt_vbpok-pstyv = 'ELP'.
APPEND lt_vbpok.
lt_repack-SOURCEHU = gs_vekp-exidv.
lt_repack-DESTHU = gs_vekp-exidv.
lt_repack-PACK_QTY = p_quan.
lt_repack-BASE_UOM = 'EA'.
lt_repack-MATERIAL = p_matnr1.
lt_repack-PLANT = p_plant.
lt_repack-DELIV_NUMB = gv_vbeln.
lt_repack-DELIV_ITEM = p_item1.
APPEND lt_repack.
CALL FUNCTION 'WS_DELIVERY_UPDATE_2'
    EXPORTING
      VBKOK_WA               = ls_vbkok_wa
      DELIVERY               = gv_vbeln
      update_picking         = 'X'
      no_messages_update_1   = 'X'
      synchron               = 'X'
      commit                 = ' '
      NICHT_SPERREN_1        = 'Y'
      if_database_update_1   = '1'
      if_error_messages_send = 'X'
    TABLES
      PROT                   = lt_prot
      VERKO_TAB              = lt_verko
      VERPO_TAB              = lt_verpo
      IT_REPACK = lt_repack.
COMMIT WORK and WAIT.
Thanks,
Shankar Darbha

Ipad shows all regular Iphone inbound calls as Facetime. I only want true FT calls to appear. I switch audio/video and it still happens. How can I set the Ipad to only register true facetime calls rather than all inbound regular calls? thanks

My Ipad will show all regular inbound calls, from Iphone, as Facetime calls. I would like only TRUE facetime calls to appear. I have switched audio/video on Ipad but it still happens. How can I set the Ipad to only register only true FT calls rather than all inbound calls that a re not facetime calls? Yes, you may have to reread this twice and slowly to understand..... Thanks!

Hi webn,
I understand that you do not wish to see phone calls on your iPad anymore. I have an article for you with some information on accomplishing this goal:
Connect your iPhone, iPad, iPod touch, and Mac using Continuity - Apple Support
https://support.apple.com/en-us/HT6337
Turn off iPhone cellular calls
To turn off iPhone cellular calls on your iPad or iPod touch, go to Settings > FaceTime and turn off iPhone Cellular Calls.
On your Mac, open the FaceTime app and go to FaceTime > Preferences. Click Settings and deselect the iPhone Cellular Calls option.
Thanks for coming to the Apple Support Communities!
Cheers,
Braden

Inbound calls to response group - failed

Hi!
I have a problem with inbound calls to Response group from external phones. But calls from Lync client to Response Group are working without problem. And calls from external phones to Lync users perfectly working too.
I found this alert in event viewer
Event 31172 LS response group service
The workflow runtime encountered an error while connecting the call.
The workflow runtime encountered a critical error.
Failure occurrences: 2, since 27/06/2013 9:20:18 PM.
The last encountered error was from a workflow having the display name: Group2, the URI: sip:[email protected], and
the GUID: 12ee27bf-15ab-4cc9-8265-57a08f17ac5b.
Exception: Microsoft.Rtc.Signaling.OperationTimeoutException - This operation has timed out.
Inner Exception: -
I hope for your help!!! Thanks!!

Hi,
Here is a similar case for your reference. Try to remove NAT and have the SIP line connect direct to Lync Server.
http://social.technet.microsoft.com/Forums/lync/en-US/d0e53ab2-42e7-4d79-be01-b8770d508133/calls-to-workflows-fail
Kent Huang
TechNet Community Support

SQL query in ART database for incoming calls

This is the sql statement I am using to pull records from ART database below. After looking at the data pulled I am not getting incoming calls. What colum do I need to include.
select user_id, orig_party_number, orig_called_party_number, final_destination, connect_date, disconnect_date, (disconnect_date - connect_date) as duration, call_type, call_classification from Tbl_Billing_Data where orig_party_number = '50656' and timestamp > '2006-01-03' and timestamp < '2006-01-31'

Query the CDR DB directly instead of ART and use this query for inbound:
PARAMETERS [Called Number] Text ( 255 ), [Start Date] DateTime, [End Date] DateTime;
SELECT DateAdd("s",[dateTimeOrigination],"31-Dec-1969 19:00:00") AS [Origin DateTime], IIf(dateTimeConnect<>0,DateAdd("s",[dateTimeConnect],"31-Dec-1969 19:00:00"),DateAdd("s",[dateTimeDisconnect],"31-Dec-1969 19:00:00")) AS ConnectDateTime, DateAdd("s",[dateTimeDisconnect],"31-Dec-1969 19:00:00") AS DisconectDateTime, dbo_CallDetailRecord.callingPartyNumber, dbo_CallDetailRecord.finalCalledPartyNumber, dbo_CallDetailRecord.duration AS Duration
FROM dbo_CallDetailRecord
WHERE (((dbo_CallDetailRecord.callingPartyNumber)<>"") And ((dbo_CallDetailRecord.finalCalledPartyNumber)=[Called Number]) And ((dbo_CallDetailRecord.dateTimeOrigination)>=DateDiff("s",#31/12/1969 19:0:0#,[Start Date]) And (dbo_CallDetailRecord.dateTimeOrigination)<=DateDiff("s",#31/12/1969 19:0:0#,[End Date])));
adignan - berbee

User Exit for Inbound Process - Changing IDOC control record

Hi,
I want to utilize the identification field (under details tab) of the IDOC Control Record. What is the suitable user exit for inbound process that i can use to change the IDOC control record?

It's been a long time since I have done any IDOC processing.
Go to SE37 and enter the function module IDOC_INPUT_ORDERS. Display the source code. Search for "CALL CUSTOMER". You should see several "CALL CUSTOMER-FUNCTION" lines. These are enhancements where you can manipulate the IDOC data. If you locate the correct one (where the data is read from the IDOC but not yet validated), you should be able to map the partner function value. Double-clicking on the number in quotes will bring you to an "EXIT_" function module. A "ZX" INCLUDE program is referenced. If you haven't used the exit yet, double-clicking on the ZX INCLUDE will create it. The values passed into the function module are available in the INCLUDE.
Ideally, you should create a project in transaction CMOD that references the chosen enhancement. This way you can activate/deactivate without commenting/uncommenting the code in the INCLUDE.

UCCX HR Report for Outbound Calls

Hi,
We are using UCCX 9.0.
In Historical report we able to get the report for outbound call also.
The issue for the Outbound is calls which are not answered by the remote party There is also a call duration though the CUCM CDR shows zero seconds and the Agent Call Records in CAD shows the call and unanswered.
Because of this we are not able to identify the matured calls & non matured calls for the outgoing.
Is there anyway to capture unanswered outbound calls from Historical reports
Thanks,
Paul

I must say that I had similar problems with reports for outbound calls on our CC system...after some writings with TAC we concluded that reports for outbound calls in CC system are much more less informational versus inbound calls. So what I want to say is that probably your reporting is working fine and there is just no info for outbound calls which can be utilized and be usefull for you in your reports...
BR,
Dragan

Transfer Order for Inbound delivery with multiple line items...

Hi All,
<The flow of my report pgm is >
PO creation using Transaction - me21n
My report pick the PO created as per date given in selection screen-
Creates outbound Deliveries using BAPI_OUTB_DELIVERY_CREATE_STO
Create Transfer order FM L_TO_CREATE_DN
Confirm Transfer order FM L_TO_CONFIRM
GI is done - Inbound delivery is created
Create Transfer order FM L_TO_CREATE_DN for Inbound delivery
Confirm Transfer order FM L_TO_CONFIRM
Goods receipt.
If PO is having one item - the entire flow is working perfectly.
if my PO is having multiple items,
While creating TO for inbound deliveeies - i am facing the error.
Before calling FM,
I am assigining these values like
wa_delit-nlpla = wa_ltap_vb-vlpla.    "Source Storage Bin To Destination Bin
      wa_delit-nltyp = wa_ltap_vb-vltyp.    "Source Storage Type To Destination Storage type
      wa_delit-anfme = wa_ltap_vb-nsolm.    "Actual destination quantity
      wa_delit-letyp = wa_ltap_vb-letyp.    "Storage Unit Type
      wa_delit-altme = wa_ltap_vb-meins.    "Unit of measure
      LOOP AT i_lips INTO wa_lips WHERE vbeln = WA_VBFA-IN_DELIV.
        wa_delit-posnr = wa_lips-posnr.
        APPEND wa_delit TO it_delit.
         ENDLOOP.
As i am appending i am getting 2 line items in the tabke it_delit.
Function module :
CALL FUNCTION 'L_TO_CREATE_DN'
EXPORTING
   i_lgnum                         = l_lgnum
   i_vbeln                         = wa_vbfa-in_deliv
   i_commit_work                   = 'X'
   i_bname                         = sy-uname
   it_delit                        = it_delit
IMPORTING
   e_tanum                         = gv_tonum_1
TABLES
   t_ltap_vb                       = i_ltap_vb_1
EXCEPTIONS
   foreign_lock                     = 1
   dn_completed                     = 2
   partial_delivery_forbidden       = 3
   xfeld_wrong                      = 4
   ldest_wrong                      = 5
   drukz_wrong                      = 6
   dn_wrong                         = 7
   squit_forbidden                  = 8
   no_to_created                    = 9
   teilk_wrong                      = 10
   update_without_commit            = 11
   no_authority                     = 12
   no_picking_allowed               = 13
   dn_hu_not_choosable              = 14
   input_error                      = 15
   error_message = 16
   OTHERS = 32.
Here i am getting as input error, i need to create only one TO for multplie items
If i pass this i a loop - then it will create 2 Transfer order which is WRONG ...so please help me to over come this error.
Thanks
Senthil
Edited by: Senthil Kumar on Dec 7, 2009 3:29 PM

Hi,
Please check FM L_TO_CREATE_MULTIPLE.
Regards,
Ferry Lianto

No Redundancy for Inbound Calls

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