VXML RINGTONE SERVICE Problem - Call is not routed to agents
We got below mentioned error in CVP Call Server logs for our calls. Due to this calls are not routed to agents.
Apart from our call flow, I need to know what is the cause and solution of this error.
Aborting XFER and disconnecting the caller code 488. RINGTONE SERVICE is not answering within 5000 millisecs, or the caller did not receive or accept the reinvite for ringtone media setup.
1646: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Processing ,, [MsgBus:CALL_STATE_EVENT], ssId=SYS_IVR1, eventId=DISCONNECT, causeCode=NORMAL_COMPLETION,, LEGID = , DNIS = 9555210577, ANI = sip:[email protected]:5060
1647: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Publishing ,, [ICM_EVENT_REPORT], dialogueId=112, sendSeqNo=2, eventId=DISCONNECT, causeCode=NORMAL_COMPLETION,, LEGID = , DNIS = 9555210577, ANI = sip:[email protected]:5060
1648: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Deleted dialogue. Duration: 0 hrs, 0 mins, 0 secs, 109 msecs
2017: 172.20.242.103: May 14 2014 11:50:07.717 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=DIALOG_CALLBACK.7} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Reinvitation proceeding TRYING.
2018: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=pool-1-thread-66-SIP-7351} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Called ring leg: CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D1273EDA6511E3999FDDDE4246E36C-140005740768575 - [RING-OUT]: status code = 0: elapsed msecs = 5000
2019: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Aborting XFER and disconnecting the caller code 488. RINGTONE SERVICE is not answering within 5000 millisecs, or the caller did not receive or accept the reinvite for ringtone media setup. (current=1 max=226) [id:5004]
2020: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=pool-1-thread-74-SIP-7355} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Waiting 2000 millisecs before terminating.
69088: 172.20.242.103: May 14 2014 11:50:12.873 +0300: %CVP_9_0_RPT-7-handleFakeNewCall: {Thrd=Thread-58} create fake New Call for >>HEADERS: (JMSType)=MsgBus:VXML_SCRIPT_DETAIL (JMSDestination)=Topic(CVP.VXMLSERVER.REPORT) (JMSTimestamp)=1400057412857 (ServerID)=cvp9lab2.SYS_VXML1:VXML:VXML1:cvp9lab2.MsgBus001 >>BODY: elementName=start elementid=1001871400057412857 timezone=Asia/Riyadh callguid=2BEF7F5ADA6511E399B0DDDE4246E36C localOffset=180 sessionname=172.20.242.103.1400057412842.21144.outbound ani=sip:172.20.243.187 howEventExited=1 sessionvars= sessionid=1001881400057412857 eventExitState=next uui=NA appName=outbound callStartDatetime=Wed May 14 11:50:12 AST 2014 elementTypeID=0 isNewCall=true vxmldatetime=Wed May 14 11:50:12 AST 2014 version=CVP_9_0 calltypeid=6 category=0 iidigits=NA dnis=sip:[email protected]:5060 >>STATE: isTabular=false isWriteable=false cursor=-1
69089: 172.20.242.103: May 14 2014 11:50:12.873 +0300: %CVP_9_0_RPT-7-handleFakeNewCall : {Thrd=Thread-58} 2BEF7F5ADA6511E399B0DDDE4246E36C onHold start time: Wed May 14 11:50:12 AST 2014
Hi,
//After that the agent goes to Reserved state but the call doesn't come through.//
I don't have experience on PCCE , but UCCE perspective this looks to me more like ATR(Agent Targeting rule) or Device Target Issue. Ringtone service should not cause Calls to fail.
Can you please post log from CVP, for Particular call that faced this issue?
Regards
Chintan
Similar Messages
-
CALL DOES NOT ROUTE OUT THE LOCAL GATEWAY
Local calls will not route out the local Gateway of branch1 to the PSTN or from the PSTN back to branch1, however they will route out either CorpHQ or branch2 backup gateways. When I go into the route group configuration for branch1, and remove the backup gateways, I get a fast busy tone when I dial the local number. I know the MGCP Gateway at branch1 is functioning because when I dial 911 and run debug ISDN Q931, the call routes properly through branch1, so I have a call routing problem. I ran DNA and it came back as ROUTE THIS PATTERN and all of the number translations looked accurate, so I didn't have to check for any block patterns. I'm not getting any errors on the calling party phone display. When I deleted the route pattern for the branch1 site and forced it to use the global route pattern, I received a debug output on branch1. I do not know a debug command (such as debug voip dial-peer or debug ccsip messages) to use for an MGCP Gateway to see if the call is actually reaching the Gateway.
I have checked the following:
the route pattern configuration
the translation pattern configuration
the called party transformation pattern configuration
the route list configuration to make sure the correct route group for branch1 was selected
the route group configuration to make sure that the branch1 Gateway was first in the order of selected devices
the route pattern configuration to make sure the correct route list for branch1 ist selected
the Gateway configuration to make sure it's using the device pool for branch1 and to make sure the called party transformation CSS for the branch1 Gateway is applied
the device pool configuration to make sure it's using the route group branch1
Any assistance would be greatly appreciated
Regards,
RonHi Nishant:
Please see the attachments for the Gateway pages
The significant digits for inbound calls for all 3 gateways is '4'
Please see the running-configs of the 3 gateways and the PSTN
Please see the debugs for the INBOUND calls
Many Thanks,
Ron
The following INBOUND call from the PSTN to 2065011001 is now working, however it is supposed to be routing through CorpHQ and is instead routing through Branch1. Please see 'DEBUG VOIP CCAPI INOUT' & 'DEBUG ISDN Q931'
Branch1#
ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0096
Cause i = 0x8290 - Normal call clearing
//22/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x16, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=22
//22/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x16, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=22
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
//22/5A001212800B/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Network, Params=0x0, Call Id=22
//23/5A001212800B/CCAPI/ccGetCallStatistics:
Call Stats=0x4A5346FC, Call Id=23
//22/5A001212800B/CCAPI/ccConferenceDestroy:
Conference Id=0xC, Tag=0x0
//22/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x49E07FD4, Source Call Id=22,
Destination Call Id=23, Disposition=0x0, Tag=0x0
//23/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/cc_generic_bridge_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//22/5A001212800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x495BABA4, Tag=0x0, Call Id=23,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DDE8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 1
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49E07FD4, Tag=0x0, Call Id=22,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DEC8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x8096
ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0096
ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0097
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Seattle US Phone'
Calling Party Number i = 0x4180, '2065015111'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xC1, '2065011001'
Plan:ISDN, Type:Subscriber(local)
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x49E07FD4, Interface Type=6, Destination=, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179280,feature_id:24
//24/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1
//24/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524790
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x495BABA4, Interface Type=9, Destination=0.0.0.0, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 2
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179056,feature_id:25
//25/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
//25/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524580
//25/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x495BABA4, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
//25/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/cc_api_call_proceeding:
Interface=0x49E07FD4, Progress Indication=NULL(0)
//24/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x49E07FD4, Data Bitmask=0x1, Progress Indication=DESTINATION IS NON ISDN(2),
Connection Handle=0
//24/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E7368, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, callID2=0x19, tag=0x0)
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, gcid=0-0-0-0, tag=0x0)
//25/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID2=0x19, gcid=0-0-0-0, tag=0x0)
//24/74820328800C/CCAPI/ccConferenceCreate:
Conference Id=0x4A2E757C, Call Id1=24, Call Id2=25, Tag=0x0
//24/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x49E07FD4, Source Call Id=24,
Destination Call Id=25, Disposition=0x0, Tag=0xFFFFFFFF
//25/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/cc_generic_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=25)
//25/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=24)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,
Modem=0x2, Codec Bytes=20, Signal Type=3)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=0x4, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=20, Signal Type=2)
//25/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Entry(Context=0x4A524790)
//24/74820328800C/CCAPI/cc_process_notify_bridge_done:
Conference Id=0xD, Call Id1=24, Call Id2=25
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms)
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=4000(ms), Initial Digit Timeout=4000(ms))
//24/74820328800C/CCAPI/ccRestartDigitTimeoutMsec:
Digit Timeout=0, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8097
Channel ID i = 0xA98381
Exclusive, Channel 1
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E6E68, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8097
Progress Ind i = 0x8088 - In-band info or appropriate now available
//24/74820328800C/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Ring Back,
Tone Direction=Network, Params=0x0, Call Id=24
//24/74820328800C/CCAPI/cc_handle_inter_digit_timer:
Generate inter-digit timeout CC_EV_CALL_DIGIT_END event
The following INBOUND call from the PSTN to 5126022001 fails and is supposed to be routing through Branch1 and is instead routing through CorpHQ. Please see 'DEBUG VOIP CCAPI INOUT'
CorpHQ#
//-1/A31ADF52800B/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=5126026222
cisco-anitype=4
cisco-aniplan=1
cisco-anipi=0
cisco-anisi=0
dest=5126022001
cisco-desttype=4
cisco-destplan=1
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
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
//-1/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Interface=0x49F42894, Call Info(
Calling Number=5126026222,(Calling Name=)(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
Incoming Dial-peer=1, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
In: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
Out: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1241383960,feature_id:13
//13/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN))
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
Event=0x497D0010
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 5126022001
//13/A31ADF52800B/CCAPI/ccCallSetContext:
Context=0x4A131A54
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 13 with tag 1 to app "_ManagedAppProcess_Default"
//13/A31ADF52800B/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
//13/A31ADF52800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49F42894, Tag=0x0, Call Id=13,
Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 49FE0410
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
PSTN#sh run
Building configuration...
Current configuration : 13975 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname PSTN
boot-start-marker
boot-end-marker
card type e1 0 0
card type t1 0 1
logging message-counter syslog
no aaa new-model
clock timezone EST -5
clock summer-time EST recurring
network-clock-participate wic 0
network-clock-participate wic 1
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
ip domain name att.com
ip name-server 177.1.100.110
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol cisco
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
header-passing
voice translation-rule 101
rule 1 /^\+.*/ //
rule 2 /^501.*/ //
rule 3 /^1206.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^206/ /1206/
rule 7 /^1512.*/ /\0/
rule 8 /^011\(.*\)/ /\1/
voice translation-rule 102
rule 1 /^1\(2065015111\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any subscriber plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any subscriber plan any isdn
rule 4 /^1\(5126026222\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^15126022.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 103
rule 1 /^206.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 201
rule 1 /^\+.*/ //
rule 2 /^602.*/ //
rule 3 /^1512.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^512/ /1&/
rule 7 /^1206.*/ /&/
rule 8 /^011\(31.*\)/ /\1/
voice translation-rule 202
rule 1 /^1\(5126026222\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any national plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any national plan any isdn
rule 4 /^1\(2065015111\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^12065011.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 203
rule 1 /^512.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 301
rule 1 /^\+.*/ //
rule 2 /^20.*/ //
rule 3 /^0\([1-8].*\)/ /31\1/
rule 4 /^011/ //
rule 5 /^0031/ //
rule 6 /^703..../ /3120&/
rule 7 /^00\(1.*\)/ /\1/
voice translation-rule 302
rule 1 /^31207037333$/ /7037333/ type any subscriber plan any isdn
rule 2 /^7033\(...\)$/ /0207033\1/
rule 3 /^911$/ /112/ type any unknown plan any unknown
rule 4 /^31\(670357575\)$/ /0\1/ type any national plan any isdn
rule 5 /^31\(107047444\)$/ /0\1/ type any national plan any isdn
rule 6 /^12065015555$/ /&/ type any international plan any isdn
rule 7 /^12065015151$/ /&/ type any international plan any isdn
rule 8 /^12065015111$/ /&/ type any international plan any isdn
rule 9 /^15126026222$/ /&/ type any international plan any isdn
rule 10 /^12065011...$/ /&/ type any unknown plan any unknown
rule 11 /^15126022...$/ /&/ type any unknown plan any unknown
rule 12 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 303
rule 1 /^703.*/ /&/ type any subscriber plan any isdn
rule 2 /^010/ // type any national plan any isdn
rule 3 /^1/ // type any international plan any isdn
voice translation-rule 1000
rule 1 /.*\(1...$\)/ /206501\1/
rule 2 /.*\(2...$\)/ /512602\1/
rule 3 /.*\(45..$\)/ /020757\1/
voice translation-rule 1001
rule 1 /^1206...5...$/ /+&/
rule 2 /^1512...6...$/ /+&/
rule 3 /^31.0...7...$/ /+&/
voice translation-profile 1-HQ-Change_DNIS-Check_ANI
translate called 101
voice translation-profile 1-HQ-Proper_Types
translate calling 102
translate called 103
voice translation-profile 2-BR1-Change_DNIS-Check_ANI
translate called 201
voice translation-profile 2-BR1-Proper_Types
translate calling 202
translate called 203
voice translation-profile 3-BR2-Change_DNIS-Check_ANI
translate called 301
voice translation-profile 3-BR2-Proper_Types
translate calling 302
translate called 303
voice translation-profile SIP-NORMALIZE-DNIS-ANI
translate calling 1001
translate called 1000
voice-card 0
dspfarm
archive
log config
hidekeys
controller E1 0/0/0
clock source internal
pri-group timeslots 1-3,16
description == Voice Circuit to Branch2
controller T1 0/1/0
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to CorpHQ
controller T1 0/1/1
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to Branch1
interface Loopback0
ip address 177.1.254.254 255.255.255.255
interface Loopback10
ip address 177.1.254.250 255.255.255.255
interface Loopback11
ip address 177.1.254.251 255.255.255.255
interface FastEthernet0/0
description ==TO INTERNET==
ip address 192.168.1.150 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description === To HQ
ip address 177.1.19.1 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:15
description == PRI Circuit to R3-BR2
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no isdn outgoing ie network-facility
isdn outgoing display-ie
no cdp enable
interface Serial0/1/0:23
description == PRI Circuit to R1-HQ
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan
isdn outgoing display-ie
no cdp enable
interface Serial0/1/1:23
description == PRI Circuit to R2-BR1
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling
isdn negotiate-bchan resend-setup
isdn outgoing ie network-facility
no cdp enable
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 3-BR2-Change_DNIS-Check_ANI
description == Voice PRI to Branch2
voice-port 0/1/0:23
translation-profile incoming 1-HQ-Change_DNIS-Check_ANI
description == Voice PRI to CorpHQ
voice-port 0/1/1:23
translation-profile incoming 2-BR1-Change_DNIS-Check_ANI
description == Voice PRI to Branch1
dial-peer voice 1 pots
description == All inbound calls from HQ BR1 BR2 into PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 101 pots
description == Subscriber Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^2065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 102 pots
description == National Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 103 pots
description == International Calls into CorpHQ from PSTN Coming from NL Ph
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^0012065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 104 pots
description == + Calls into CorpHQ from PSTN Coming from Mobiles
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern +12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 201 pots
description == Subscriber Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^5126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 202 pots
description == National Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 203 pots
description == International Calls into Branch1 from PSTN Coming from NL Ph
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^0015126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 204 pots
description == + Calls into Branch1 from PSTN Coming from Mobiles
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern +15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 301 pots
description == Subscriber Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^7033...$
direct-inward-dial
port 0/0/0:15
forward-digits 7
dial-peer voice 302 pots
description == National Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^0207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 10
dial-peer voice 303 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^01131207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 304 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 305 pots
description == + Calls into Branch2 from PSTN Coming from Mobiles
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern +31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 1000 voip
description == Calls into AT&T SIP ITSP for VC Week1 Lab1
rtp payload-type nse 99
rtp payload-type nte 100
voice-class sip localhost dns:sip1.att.com
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 5000 voip
service aa
destination-pattern A5000
session target ipv4:177.1.254.254
incoming called-number A5000
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
num-exp 1888....... 911
num-exp 1900....... 911
num-exp 1976....... 911
num-exp 1777....... 911
num-exp 1444....... 911
num-exp 0800....... 911
num-exp 0900....... 911
sip-ua
telephony-service
no auto-reg-ephone
max-ephones 1
max-dn 10
ip source-address 177.1.254.254 port 2000
caller-id block code *67
system message You WILL PASS this Exam!
voicemail A5000
max-conferences 8 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Sep 01 2012 15:29:37
ephone-dn 1 dual-line
number 12065015111 secondary +12065015111
label Seattle, US +1 206 501 5111
description INE PSTN Phone
name Seattle US Phone
ephone-dn 2 dual-line
number 15126026222 secondary +15126026222
label Austin, US +1 512 602 6222
name Austin TX Phone
ephone-dn 3 dual-line
number 31207037333 secondary +31207037333
label Amsterdam, NL +31 20 703 73 33
name Amsterdam NL Phone
ephone-dn 4 dual-line
number 12065015555 secondary +12065015555
label Hurley Mobile +1 206 501 5555
name Hurley's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 5 dual-line
number 12065015151 secondary +12065015151
label Hurley's Home +1 206 501 5151
name Hurley's Home
call-forward busy A5000
call-forward noan A5000 timeout 12
ephone-dn 6 dual-line
number 31670357575 secondary +31670357575
label Sawyer's Mobile +31 6 70357575
name Sawyer's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 7 dual-line
number 911 secondary 112
label US/EU Emer/FreePhone/Prem
name Emergency Services
ephone-dn 8 dual-line
number 15126026262 secondary +15126026262
label BLinus Mobile +1 512 602 6262
name Benjamin Linus Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 9 dual-line
number 31207037373 secondary +31207037373
label DHume Home +31 20 703 73 73
name Desmond Hume Home
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 10 dual-line
number 31107047444 secondary +31107047444
label Rotterdam, NL +31 10 704 74 44
name Rotterdam NL Phone
ephone 1
device-security-mode none
mac-address A456.3040.0DAA
type 7975
button 1:1 2:2 3:3 4:10
button 5:6 6o7,8,5,4
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 10
ntp server 64.90.182.55
end
CorpHQ#sh run
Building configuration...
Current configuration : 6353 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname CorpHQ
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone PST -8
clock summer-time PDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 177.1.11.1 177.1.11.14
ip dhcp excluded-address 177.1.11.21 177.1.11.254
ip dhcp excluded-address 177.2.11.1 177.2.11.14
ip dhcp excluded-address 177.2.11.21 177.2.11.254
ip dhcp pool CorpHQ-Phones
network 177.1.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.1.11.1
dns-server 177.1.100.110
ip dhcp pool Branch1-Phones
network 177.2.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.2.11.1
dns-server 177.1.100.110
no ip domain lookup
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
fax protocol cisco
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice translation-rule 1
rule 1 // // type any subscriber plan any isdn
voice translation-rule 2
rule 1 // // type any national plan any isdn
voice translation-rule 3
rule 1 // // type any international plan any isdn
voice translation-rule 10
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011/ /9&/
voice translation-profile MakeInternational
translate called 3
voice translation-profile MakeNational
translate called 2
voice translation-profile MakeSubscriber
translate called 1
voice translation-profile Prefix9_InFrom_CUCM
translate called 10
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24
description == Voice Circuit to PSTN
interface Loopback0
ip address 177.1.254.1 255.255.255.255
ip pim dense-mode
interface FastEthernet0/0
description == To CorpHQ-Switch
no ip address
duplex auto
speed auto
interface FastEthernet0/0.10
description == Server VLAN
encapsulation dot1Q 10
ip address 177.1.10.1 255.255.255.0
ip pim dense-mode
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.1.11.1 255.255.255.0
ip helper-address 177.1.10.10
ip nbar protocol-discovery
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.1.12.1 255.255.255.0
interface FastEthernet0/0.13
description == PSTN PHONE VLAN
encapsulation dot1Q 13
ip address 177.1.13.1 255.255.255.0
interface FastEthernet0/1
description === To PSTN
ip address 177.1.19.254 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 36
cdp enable
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To BR1
bandwidth 384
ip address 177.0.101.1 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/0.2 point-to-point
description == FR To BR2
ip address 177.0.201.1 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 177.1.19.1
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0.10
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
voice-port 0/3/0
voice-port 0/3/1
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 1 register CorpHQ-729-MTP
associate profile 2 register CorpHQ-711-MTP
associate profile 3 register CorpHQ-HW-Xcode
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
translation-profile outgoing MakeSubscriber
destination-pattern 911
no digit-strip
port 0/0/0:23
dial-peer voice 11 pots
translation-profile outgoing MakeSubscriber
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
dial-peer voice 12 pots
translation-profile outgoing MakeNational
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
dial-peer voice 13 pots
translation-profile outgoing MakeInternational
destination-pattern 9011T
port 0/0/0:23
prefix 011
dial-peer voice 100 voip
description == Inbound/Outbound SIP PSTN GW From/To CUCM Pub
translation-profile incoming Prefix9_InFrom_CUCM
destination-pattern ^2065011...$
voice-class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer hunt 1
sip-ua
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 2
ntp server 177.1.254.254
end
Branch1#sh run
Building configuration...
Current configuration : 3838 bytes
! Last configuration change at 01:19:02 CDT Thu Oct 10 2013
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch1
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone CST -6
clock summer-time CDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip multicast-routing
no ipv6 cef
ntp update-calendar
ntp server 177.1.254.1
multilink bundle-name authenticated
isdn switch-type primary-ni
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24 service mgcp
interface Loopback0
ip address 177.1.254.2 255.255.255.255
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.2.11.1 255.255.255.0
ip helper-address 177.1.254.1
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.2.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn supp-service name calling
isdn bind-l3 ccm-manager
isdn outgoing ie facility
isdn outgoing display-ie
isdn outgoing ie redirecting-number
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.101.2 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
ccm-manager fallback-mgcp
ccm-manager redundant-host 177.1.10.10
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
mgcp
mgcp call-agent 177.1.10.20 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp profile default
sccp local Loopback0
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 3 register Br1-HW-Xcode
associate profile 1 register Br1-729-MTP
associate profile 2 register Br1-711-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
end
Branch2#sh run
Building configuration...
Current configuration : 5789 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch2
boot-start-marker
boot system flash:c2800nm-advipservicesk9-mz.124-24.T7.bin
boot system flash:
boot-end-marker
card type e1 0 0
logging message-counter syslog
no aaa new-model
clock timezone CEST 1
clock summer-time CEDT recurring
network-clock-participate wic 0
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
no supplementary-service h225-notify cid-update
fax protocol cisco
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class custom-cptone JOIN-TONE
dualtone conference
frequency 300 3600
cadence 150 100 500
voice class custom-cptone LEAVE-TONE
dualtone conference
frequency 300 3600
cadence 500 100 150
voice translation-rule 1
rule 1 /^7033...$/ /020&/
voice translation-rule 10
rule 1 /^0/ /0&/
voice translation-rule 200
rule 1 /^206501...$/ /1&/
voice translation-profile 7DigitDNIS-to-10Digit
translate called 1
voice translation-profile Prefix0_InFrom_CUCM
translate called 10
voice translation-profile Prefix1-toCorpHQ-ANI
translate calling 200
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller E1 0/0/0
pri-group timeslots 1-3,16
description == Voice Circuit to PSTN
controller E1 0/0/1
interface Loopback0
ip address 177.1.254.3 255.255.255.255
h323-gateway voip bind srcaddr 177.1.254.3
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
encapsulation dot1Q 11
ip address 177.3.11.1 255.255.255.0
ip helper-address 177.1.10.10
interface FastEthernet0/0.12
encapsulation dot1Q 12
ip address 177.3.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.201.2 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
interface Service-Engine1/0
no ip address
shutdown
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 7DigitDNIS-to-10Digit
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 4 register Br2-HW-Conf
associate profile 3 register Br2-HW-Xcode
associate profile 2 register Br2-711-MTP
associate profile 1 register Br2-729-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 4 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
conference-join custom-cptone JOIN-TONE
conference-leave custom-cptone LEAVE-TONE
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
destination-pattern 112
no digit-strip
port 0/0/0:15
dial-peer voice 11 pots
destination-pattern 00[1-9]T
port 0/0/0:15
prefix 0
dial-peer voice 12 pots
translation-profile outgoing Prefix1-toCorpHQ-ANI
destination-pattern 000T
port 0/0/0:15
prefix 00
dial-peer voice 100 voip
description == Inbound/Outbound H323 PSTN GW From/To GK and CUCM Pub
translation-profile incoming Prefix0_InFrom_CUCM
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer voice 101 voip
description == Outbound H323 PSTN GW To CUCM Sub
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.20
ip qos dscp cs3 signaling
dial-peer hunt 1
telephony-service
max-ephones 1
max-dn 1
ip source-address 177.1.254.3 port 2000
max-conferences 8 gain -6
moh test.au
multicast moh 239.2.1.1 port 16384 route 177.1.254.3 177.3.11.1
transfer-system full-consult
create cnf-files version-stamp 7960 Sep 13 2013 18:55:27
line con 0
exec-timeout 0 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
login
scheduler allocate 20000 1000
ntp source Loopback0
ntp update-calendar
ntp server 177.1.254.1
end -
Iphone 4S; Phone & audio piece; Incoming calls are NOT routed automatically??
When I dial out; no problem the calls automatically goes through ear pice.
When I get incomming calls the call goes to phone, not even to the speaker in in the phone. So I have to
1.Slide the bar (then first realizing (again) dam: I hear nothing)
2. Choose the box for speaker options
3. Choose my blue ant piece..
By then some callers think I am not picking up or there is somthing wrong..!
My 3 colleaguers have same phone and no such problems. We are at a loss and I could not find reply in community groups. There was note about Iphone 5 and not correcting to Car speaker automatically. This was noted as a softwre issue Apple might address in next version. But since my colleagues do not have problem with their Iphone it must be either my settings or a weird fault with my phone.
One of my collegues have same ear pice as me, so the issueis not that either.
Is there some weird setting I need to fix to correct the problem?
Thank youAre you able to make calls? I would try a couple different things -
1. Try doing a double hard reset - Press and hold home button and the sleep wake button at the same time until the screen displays the slide to shut down red button. Continue pressing and holding both the buttons until the screen goes black and then the apple logo flashes up. Continue pressing and holding both the home and sleep wake button beyond this point until the screen goes black again. At this point, let go of the home button and press once the sleep wake button as you would to start your phone. See if this double hard reset helps clear out any cache.
2. If this fails, I would try restoring the phone as a new phone in iTunes. Backup your phone before this so that you dont lose your contacts, etc. You can do a couple test calls once the firmware is installed before setting up the phone, data, apps, etc. If it looks good, try restoring your phone from an earlier backup.
If everything fails, call AppleCare or see a Genius in the store. -
Web Service : problem calling it from PDF.
Hi everyone,
I want to make a call to a web service created by SAP/ABAP.
I have the wsdl file and I created a new data connection in the Adbobe designer.
I draged and droped the data connection to the editor : so it created
2 text fields and a button.
The web service waits from the user to enter an input string and it gives back as an answer a string as output...simple, isn't it ?
I also added a button in the designer that points to the connection I created. The button is in mode "execute".
Finally, I saved all this in a PDF file.
When I double click the PDF file, I enter a string as input and press the button....Then I go an error message saying that I should specify username and password. So I changed the mode of the button from "Execute" to "Submit". I added to the URL my username and password because the web service points to a backend which is SAP.
Now I do get another error message which is : Cannot handle content type:text/xml; charset=utf-8.
I read in the doc that maybe I should use XDP ?
Can someone please explane me how to solve the problem.
Thanks in advance for the precious help.
Thanks in advance.
Regards.The problem is caused by the fact that your web service is returning a content-type header of text/html. The reader product can only handle PDF responses from a web server. In any case, your form data is received OK by the web service; it is just the response back that Acrobat cannot handle.
To get around this, I created a little "thank you for your response" PDF file and put it on my web server. I then set my perl CGI backend script to send a simple redirect to the PDF when it gets a response. (Obviously, you would want to harvest the form values from your user before redirecting). This solved the problem for me.
My little perl script is shown below:
#! /usr/bin/perl
use CGI;
$query = CGI::new(); # invoke a new CGI object for the input
@params = $query->param();
# do something with the values....
print $query->redirect(-uri => 'http://mycompany.com/myapp/cgi-bin/thankyou.pdf'); -
Call do not route to operator when users do not press any key
We have a new CUCM platform with Cisco Unity ver 9.1.2TT1.11900-2TT1.
We configured system handlers and and all "caller input" options runs good but when callers do not press any key the call goes to "nowhere", instead of going to Operator (default behavior).
Do you have any idea on what could be misconfigured here, or what should be configured to solve this problem??
Thanks a lot
EnriqueSorry for my last "wrong answer" .....You (again) was right, I had a bad configuration on "after greeting" setting..i had:
"Call Action: Take message".
I changed to " User with Mailbox pointing to recepcionist VM.
Jaime thanks a lot.
Enrique Villasana -
Service problem check link not working
Hi everyone,
For the last few days I have been trying to use the link to check if there are any known broadband problems in my area. Whenever I enter my phone number I get the message 'please enter a valid BT phone number...' Is there any other way to check this?
Cheers,
Alan.Hello,
There is a problem with the personalised checker affecting some customers, and it will be back up and running for everyone by tomorrow morning.
If after that time it's still down for you please let me know.
Thanks,
Stephanie
Stephanie
BTCare Community Manager
If you like a post, or want to say thanks for a helpful answer, please click on the Ratings star on the left-hand side of the post. If someone answers your question correctly please let other members know by clicking on ’Mark as Accepted Solution’. -
Workflow not routing to agent specified in partner profile
have included below notes regarding the mis-routing of workflow messages of "translation error" STATUS message idocs that are uploaded into R/3.
The testing that I have been performing has been in Development System. The first thing that I check is that the Partner profile parameters are set to send the STATUS message type idoc to an agent via t-code we20. This agent does exist in the organization structure found at t-code ppome. I have been using the logical system (partner type LS) user=GEIS, inbound parameter message type=STATUS, message variant=AP, where myself is setup as the agent to receive workflow.
When processing an inbound STATUS idoc containing an error code of "05", meaning that translation failed on the outbound corresponding idoc (e.g. invoice, desadv, etc.) in the EC subsystem, workflow should be triggered and route the message to the agent specified at the message type level of the partner profile. In the event that an agent is not configured at the message type level of the partner profile, then the workflow message should be routed to the agent specified at the partner profile global level. However, in the event that an agent is not specified at the message type level or partner profile global level, then the workflow message should be routed to the IDOC Administrator, which is configured in t-code we46.
The results of my testing has been that regardless of the agent configuration set at the message type level or partner profile global level, all workflow messages are routed to the IDOC Administrator configured in we46.
The request is to have development work completed so that error STATUS idocs can trigger workflow messages that will be routed to various agents for resolution, versus workflow messages being routed just to the IDOC Administrator - which is how it is working today.
I would greatly appreciate for your immediate response.
Regards,
ShankarHi Stacey
If DEV is working and PRD is not have you gone through and compared both and ensure latest MSMP configuration in PRD has been activated?
Also, is the approver COCHGG00 also the Role Owner?
Are you able to show you MSMP configuration? It's makes sense to analyse the log in the context of your configuration. E.g. does the Z_ADDTNL_ACCESS_PATH path have two stages: Manager and Role Owner of which there is a routing rule on the Manager approval to go to the NO_ROLE_OWNER path where the business role has no role owner?
Regards
Colleen -
Agent state changed to reserved but call is not ringing/landing for 30 seconds
Hi All,
we have IPCC 8.5, CVP 8.5, UCM 8.5, last few weeks we are facing
agents are facing intermittently, their state changed to reserved but call is not landing/ring for a while, and we have seen call is going to RONA in CVP logs.
We have cross checked the Device Target (4 CVP servers) its fine, Queue music is interruptable.
In Ingress/VXML gateway we have the dial-peer pointing to two subscribers with equal priority. Seems to be some call is not routing to agent phone (delay is there between voice gateway and ip phone) due to some reason. We dont use SIP proxy we use static routing to subscribers.
Please share your ideas.
with Regards,
ManivannaIn Ingress/VXML gateway we have the dial-peer pointing to two subscribers with equal priority.
The gateway should point to the Call Servers. The Call Servers should have static routes to the subscribers.
If the call is not getting to the agent even though they go into Reserved (the Call Router has selected them), ensure that the SIP trunks to the Call Servers and the agent phones are in compatible partitions/CSS. Examine the logs on the Call Server when the INVITE is sent to the agent phone. If the INVITE returns 404 (not found) or 503 (unavailable), then the setup is wrong.
Regards,
Geoff -
OSB Dynamic service call or dynamic route and transactions
Hello,
I've got the following problem in my OSB architecture for asynchronous incomming services.
OSB version is 10.3.1
ConnectionFactory is XA
Problem domain:
Dynamic service call or dynamic route within transaction boundary
Elements within transaction:
1. JMS Queue ->
2. Upper Proxy Service (De-queue) ->
3. Lower Proxy Service (Canonical Service) ->
4. Business Service (Data Service) ->
5. Database Adapter->
6. Database
Dynamic element
Upper Proxy Service (De-Queue) must be able to determine which Lower Proxy Service (Canonical Service) has to be called on the basis of the message taken from the JMS Queue without losing the transaction (this doesn’t work yet).
Already tried possible solutions:
• Make a service call dynamic by setting the $inbound name attribute. This results in a this property is read-only error.
• Use a Dynamic route instead of a service-call. This results in a loss of transaction. The message is deleted from the Queue without taking failure or success into account.
• Use a Dynamic route with a quality of service set to exactly once in the header. This seems to be ignored, results in a loss of transaction.
• Use a Dynamic route with an explicit reply with failure for errors. Results in a loss of transaction.
• Use a Dynamic Publish. This doesn’t fit the required flow because it has no response.
I hope someone can shed some light on this because I'm at a total loss. I found some references in the forum to what seems to be the same problem but without definite answers. A work-around is also a viable answer.
Regards,
Bas MulHi,
I'll try to clarify.
Upper proxy dequeues a message.
Depending on a value of a specific tag within the message a canonical proxy service (out of 5 at the moment but this numer could increase) has to be called (and be dynamic transform transformed to the format of the specific canonical).
The canonical service must then pass the message to a database adaper's business service which is responsible for the database insert/update/read.
If something in the chain from dequeue to database action goes wrong the message may not dissapear from the queue.
Somehow a dynamic route in the message flow between dequeue proxy and canonical proxy breaks the transaction.
Bas -
I am experiencing this error with one of our cluster environment. Can anyone help me in this issue.
The Cluster Service function call 'ClusterResourceControl' failed with error code '1008(An attempt was made to reference a token that does not exist.)' while verifying the file path. Verify that your failover cluster is configured properly.
Thanks,
Venu S.
Venugopal S ----------------------------------------------------------- Please click the Mark as Answer button if a post solves your problem!Hi Venu S,
Based on my research, you might encounter a known issue, please try the hotfix in this KB:
http://support.microsoft.com/kb/928385
Meanwhile since there is less information about this issue, before further investigation, please provide us the following information:
The version of Windows Server you are using
The result of SELECT @@VERSION
The scenario when you get this error
If anything is unclear, please let me know.
Regards,
Tom Li -
Router calls offered and Service Level calls offered differences
Hi,
I was checking the definition of router calls offered and service level calls offered. Could anyone please let us know the exact difference between these fields in t_Skill_group_interval table.
Regards
DeepakHi Jameson,
Thanks for the reply.
I went through the database schema .I have some doubts regarding the terminologies.
1. The component of redirect calls - Would like to know what it would be .
2. The component - still unanswered within threshold . Would this mean the calls queued longer than SL .which means all the remaining calls which were offered to this skillgroup. Hence would like to know if the Service level calls offered should be equal to Router calls offered.
3. I observe that Service level calls offered were less than the calls answered field. Would like to know the possible reasons. I was initially thinking that there are might be calls overlapping but its not so as per my testing. Please help.
Testing performed :
Added to this , there is one more field called "Calls offered" and i observe that this is incremented when the calls are answered.
I performed to test "Service Level calls offered, calls offered , Calls answered and Router calls offered" .
I made a test call and queued skillgroup at 2:27 pm and made to be in queue and answered in next interval 2:32 pm (crossing reporting interval)
I observe the below,
In 2:15 interval for the skillgroup ,
Router calls offered = 1 , Calls offered = 0 , Calls answered = 0 , Service Level calls offered = 0
In 2:30 interval for the skillgroup,
Router calls offered = 0 , Calls offered = 1, Calls answered = 1 , Service Level calls offered = 1
Would like to know if this are correct values. -
Performance problem calling service controller
Hi,
first of all, I'm a really Web Dynpro Newbie, so please be careful with me
My problem is following. I'm trying to boost performance of our business Web Dynpro project. The starting time is very high. The first view, which is displayed, calls a method of a service controller which just returns a string. With logging I found out, that between the calling of the service controller with "wdThis.wdGetXXXServiceController().getSomething()" and the wdDoInit() of the "XXXService", there is a delay of 5-10 seconds.
So, i have no idea, what is done in this seconds. The wdDoInit should the first code be called, or not?
Maybe someone has one idea. I'm afraid, that it will not be possible to post lot more of code, because I cannot locate the problem.hi
the code generated automatically when you use service controller , gets placed in
component controller init() and a execute method gets created , so its not mandotary that
you write the code in the init() , you can even write the code in the execute create , if so you seem that there delay in the time using service controller , you can even have your manual code written .
I guess there should not be any performance criteria coming in picture just because of using service controller , and is you find that , write the code . manually . -
Inter-Trunk not route incoming calls from out
Hi,
I setup one extra gateway where I try to route part of our calls. So far I have success to route internal calls into there, but when I'm making a test call from outside that ends into "number is not used" problem.
I have:
- Route ready, elsewere the internal calls are not working.
- PSTN usage, linked to the Route
- Trunk configuration where I have selected the PSTN usage
- Incoming numbers are coming in E164 format
I have also tested the "Test-CsInterTrunkRouting" and that gives "pass":
FirstMatchingRoute : Description=;NumberPattern=^\+358123654789;Name=Test
Gateway;SuppressCallerId=False;AlternateCallerId=
MatchingUsage : Test PSTN Usage
MatchingRoutes : {Description=;NumberPattern=^\+358123654789;Name=Test
Gateway;SuppressCallerId=False;AlternateCallerId=}
But still, when I made a call from outsited the OCSLogger shows that mediation server try to offer call to Front-End which says only: "SIP/2.0 404 Not Found" and then bye-bye.
What is the missing magic, which made the mediation server to see alternative route? I hope it is not required that mediation server must be collocated on the Front Ends, as that one I do not have.
Any good ideas?
ps.
I'm not sure does it matter, but my Lync gives "SIP/2.0 403 Forbidden" when there is coming call from extra gateway. But as the calls into there works, then I don't see why external calls should not also work.
PetriCould it be even so, that intra-trunk routing requires consolidated mediation server? As the call is owned by the Mediation server (stanalone), and it is trying to offer that to FE. FE reply "does not exist". Because of the standalone Mediation
server does not have the call routing engine like FE have, the call is lost.
I started to think above as Lync users are able to call to that number. So FE is able to do the routing and get calls into the correct place.
I have to say also, I have read
Ken's blog about inter-trunk routing, I have to say that I'm not so sure what he means by this: "Fortunately, in most cases, adding PSTN usages to the trunk has no effect, since there is almost always a Lync user assigned to the incoming phone
numbers". Why to add additional routing for the numbers which are already inuse? I hope it is not required, that you need to have a users ID for each number you do the inter-trunk routing?
Petri -
Well my iphone 5 does not remember my home network. This is not a problem with the wifi router as the ipad works absolutely fine. The wifi loses it connection in just 2 minutes. So could you please fix this problem.
Not unless the modem is causing a problem.
What you want to do is get it to work reliably over Ethernet first, then tackle wifi. Power off the modem. On your macbook, delete the Ethernet configuration and the Wifi configuration. Power up the modem, then connect the mac via Ethernet. Create the new Ethernet configuration and see if you can connect. -
Problem with call forwarding. Calls can not be forwarded for incoming external calls
Hi Everybody, how are you?
I have a problem with call forwarding. Everything was fine but now is not working.
In the reception of an office, the receptionist activate the call forward option to an internal extension. If somebody, internal in the office, call to the reception, the call is forwarding to the extension configured. But if I call from the outside (in example, from my cellphone) the call is not forwarded to the extension configured and continue ringing in the reception phone. Why this behavior? Any idea?
If you know something please tell me.
Thanks. Best regards.
Andres Collazos.I encounter a similar problem with 9.1.1.
My problem is link to this bug ID : CSCtq10477.
Mathieu
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its in the heading if anyone can help thanks
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Dotted Strokes not exporting to pdf (InDesign CS3).
I'm producing a booklet in CS3 where I've used a lot of dotted strokes (both as decorative lines, and as outlines to background panels beneath text). When I export to pdf, I'm finding that all the spreads that contain these elements come up blank wit
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i have a the code below which i am passing a form value to <CFLOCATION url="CalendarCurrent.cfm?DATE=<cfoutput>#form.Diary_Date#</cfoutput>"> but i am getting cannot convert to a date error, i cannot change the code below as other pages use it, so ho
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Use of ThreadLocal in EJB to store Transaction id
Hi , I my application we need to retain the transaction id for logging . We had a facade layer which are stateless session beans . We use an EJB interceptor to get the transaction id from the request object and put it in a threadlocal . We retrive th
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Vendor Total Block 99 , was unable to block against payment
Dear Friends, Vendor was total block 99 on Dec 2012. However, the system did not stop when payment was being made to the vendor on Jan, 13? Please explain when this is happening even after vendor block was done. Regards Sridhar