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
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CUBE- one way audio intermittent for inbound calls
Have a CUBE on 3800 with 15.14M(6) with SIP to SIP dial peers. Some external inbound callers at times can hear no audio streaming from IVR.
I run a traffic capture from the LAN side of the CUBE and I can hear the audio.
I run a traffic capture on the WAN side (towards caller) and i hear nothing, BUT the rtp stream is active (50 pps G711a). When i replay this stream on Wireshark i hear silence exactly what the caller can hear...
G711a on both sides but no audio is heard on the WAN side exiting the CUBE towards telco
DIal peers matched= 31 In from itsp and 40 Out to CUCM
Setup:
ITSP--SIP Trunk-- MyCUBE--SIP Trunk--CUCM--IVR(CTI Ports and RPs)
|
7900 Type A sccp Handsets
Config attachedhi,
Yes my itsp can connect and see that interface. Most calls work and only some fail. Intermittent. I have incuded a "show call active voice" for a "failed" incoming call from 0204002244 to 0498870440 . What happens here is silence to external caller coming in on dial peer 31 but there is an RTP stream being sent to him. Wireshark trace on WAN side proves silence within RTP stream exiting CUBE towards callers SBC at 10.193.202.100 from my cube at 10.66.41.1
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
ChargedUnits=0
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)
CPA Active Call Parameters
CPA Event Status: DISABLE
GENERIC:
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
AlertTimepoint=12:24:36.004 AEDT Thu Jan 30 2014
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=
Username=0204002244
MlppServiceDomainNW=0 (none)
MlppServiceDomainID=
PrecedenceLevel=0 (PRECEDENCE_LEVEL_NONE) -
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 -
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"
whenever i am changing to "H225 Info for call Progress Tone" then the branches problem getting solved but Headoffice getting the same problem.
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
h323-gateway voip bind srcaddr 192.168.21.5
interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
router eigrp 200
network 10.10.0.0 0.0.0.3
network 192.168.20.0
network 192.168.21.0
no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
neighbor 172.31.217.201 remote-as 65000
no auto-summary
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.20.1
ip route 192.168.20.50 255.255.255.255 192.168.20.1
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
access-list 23 permit 10.10.10.0 0.0.0.7
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
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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
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Edited by: [email protected] on Jun 9, 2010 12:45 AM
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