Calls disconnect after error %CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH

Hi,
If anyone could help or advise me on the following problem.
We use Cisco7206VXR for our voice gateway with PA-VXC-2TE1.
A customer is arrempting to originate calls over IP via a Qiuintum DXC2030. Rhe problem is that the calls is abruptly terminated without being able to establish the Telphony call leg.
I turn on "debug voice ccapi inout" and the following error is seen:
Error Message
%CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH
Cisco.com says reference this error:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123sup/123sems/123semv1/emgc6msf.htm#1034779
Error Message
%CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH : The called number numbering type did not matchThe event for the callid [dec] is being discarded.\n
Explanation: The event points to a session application that no longer exists. The event is being discarded.
Recommended Action: This message is informational. No action is required.
Can someone please help me to know why this is happening and how I may overcome this!
Anyone help would be greatly appreciated!
Kindest Regards
Darren Frowen
Senior Network Engineer
Satellite Media Services.

Hi GS,
Thankyou for your time and update. Then Called and calling numbers are as follows:
Called: 00441788523003
Calling: 12345676
Please see attached "show run" and Network diagram.
Your help in this matter would be gratley appreciated.
Kindest Regards
Darren Frowen

Similar Messages

FXO calls disconnect when answered

Hi
I've been struggling with this issue for over a month and both I and TAC seem to be stumped.
I have a customer with a small office that we are trying to migrate to a centralized CCM cluster. They have a 1760V with 4 analog lines that we want to connect to an FXO card. When we connect the lines to the router for testing I get the following behavior.
PSTN caller calls into the system.
The IP phone user hears the phone ring and answers the call.
The IP phone user sees the call disconnect After a second or two, the call begins to ring again on the IP phone If the IP phone user answers the call again, the call disconnects and the process repeats The PSTN caller hears ringback the whole time.
The problem seems to be that the router is interpreting a disconnect coming from the telco.
Trace snippet:-
008192: .Jul 31 08:36:28.484: htsp_process_event: [2/0,
FXOLS_PROCEEDING,
E_HTSP_CONNECT]fxols_offhook_connect
008193: .Jul 31 08:36:28.488: [2/0] set signal state = 0xC timestamp = 0
008194: .Jul 31 08:36:28.488: htsp_timer_stop
008195: .Jul 31 08:36:28.488: htsp_process_event: [2/0, FXOLS_CONNECT,
E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
008196: .Jul 31 08:36:28.716: htsp_process_event: [2/0, FXOLS_CONNECT,
E_DSP_SIG_0100]fxols_normal_battery
008197: .Jul 31 08:36:28.744: htsp_process_event: [2/0, FXOLS_CONNECT,
E_DSP_SIG_1100]fxols_offhook_disc
I've talked to the CO guy but he's been little help. I did have them run a ground start trunk for testing but it exibits the same behavior.
I've tried every combination of battery-reversal supervisory disconnect etc... that I can think of but I cannot seem to solve this problem.
TAC does not think its hardware related but I'm going to send a replacement router out just in case.
I'm hoping someone might have some suggestions for me.
Thanks in advanced.

I have seen this problem before where the polarity on the wires were reversed and/or the physical wiring is physically 4 pairs instead of 2. The physical wires going into the fxo ports should be 2 wire.
Please have the telco verify that the polarity is not reversed [ make sure that verify/re-verify ] as sometimes they claim to check, but are not diligent enough when checking.
If the gateway is somehow detecting battery-reversal, then it will essentially disconnect the PSTN call leg, which will cause the ip phone to see a disconnect. Next, because the PSTN continues to ring the call, a second call will come in [ from the view of the ip phone ]. To test if battery reversal is causing the problem, you can temporarily disable battery-reversal detection on the voice port by using the following command:
router#test voice port 1/1/0 detector battery-reversal
If after disabling battery-reversal, and this problem dissapears, then there is your problem. Please note that the test voice port command only lasts until the router is rebooted.
HTH,
Tony

Skype calls disconnection

Dear all,
I purchased skype voip services, and trying to call pakistan from saudi arabia, but during last 4 days my all calls about 30-40 calls disconnect after 1 or one and half minutes. I am using nexus 4 android 4.4.4 and skype latest version. Also skype is not blocked in saudia arabia. Also i tried to make a call from my laptop but same issue. Is there any skype customer support number or email who can check my account and solve this issue

I have the same problem with Skype calls from Saudi Arabia to landlines. Strangely, when I call from Saudi to the U.K. or Saudi to Switzerland I get clean, uninterupted phone calls. When I call Saudi to the United States the calls are disconnected after a bit more than one minute. I suspect the local telecom is doing this to discourage Skype users. Does anyone have the same experience making Skype calls from the Kingdom?

Error -63043: FPGA disconnects after running for a few seconds

Hello,
I face the following problem whenever I try to run a FPGA VI directly from my host PC onto my PXI 7831R through the ethernet.
Initially, MAX shows my remote PXI system as 'Connected', and the FPGA VI compiles fine.
But when I hit the 'Run' button, the VI seems to run fine for 3-4 seconds, after which it hangs (screenshot attached). A couple of seconds later, it returns the following message :-
" LabVIEW FPGA called another software component and that component returned the following error
Error code: -63043
NI-RIO FPGA Communications framework: The session is invalid. The target may have reset or been rebooted. Check the network connection and retry the operation. "
The VI continues to run on the FPGA, but I can't control it anymore from my host-PC. Even MAX shows the remote system as disconnected, after this error occurs. All I am left to do is to reboot my host-PC as well as the remote PXI target; to reach the beginning of this problem.
I have checked the NI-RIO settings. Both my host-PC as well as remote PXI system have the same NI-RIO 3.0.0 version installed. I am running LabVIEW 8.6 .
Can anyone help ?
Attachments:
Error Screenshot.JPG ‏89 KB

Hi everyone, I have the same problem regarding the 63043 error code
I´ve simplified my FPGA vi up to a simple write and read program of a port, as shown in the attachment, but I still have the same error.
It´s interesting that the program runs, note in the jpg file that the while loop counter suddenly stops at Numeric=10, it means the program ran 10 iterations. Also the input port succesffully read the pin value, this meas that the program ran properly, but the SBRIO rebooted when Numeric was 10 and the target device disconnected from the project, i wonder why?
The target device is an sbRIO-9632 , serial number 01452AC3 with the following software installed:
NI-RIO 3.4.0 - January 2010
DataSocket for LabVIEW Real-Time 4.7.0
EPICS Server I/O Server 1.6.1
LabVIEW PID and Fuzzy Logic Toolkit 9.0.0
LabVIEW Real-Time 9.0.1
Language Support for Japanese 9.0.1
Language Support for LabVIEW RT 1.0.0.3
Modbus I/O Server 1.6.1
NI-RIO 3.4.0
NI-Serial RT 3.3.4
NI-VISA 4.6
NI-VISA Server 4.6
NI-Watchdog 4.0.1
Network Variable Engine 1.6.1
Run-Time Engine for Web Services 2.0.1
SSL Support for LabVIEW RT 2.0.1
System State Publisher 1.1.0
Variable Client Support for LabVIEW RT 1.6.1
Web Server for LabVIEW RT 2.0.0
Available add-ons:
LabVIEW Control Design and Simulation 9.0.1
However, if I run a program in the RT processor without referencing any vi on the FPGA it runs without problems, it´s only when a file is loaded into the FPGA that the system reboots and the error message appears.
Please help me
leito
Attachments:
error-63043.jpg ‏264 KB

Free jabber disconnect after call for 10-20 sec

I try to use Free Jabber call to my system via DNS SRV record. It can connect to system but It will disconnect after call for 10-20 sec.
All case which call via DNS SRV record, I checked and found
-The problem happen only call to TPS (via conductor).
-If call to endpoint, It's working fine.
-If call to MCU (registered with VCS), It's working fine.
I see log files which new INVITE is sent by TPS to VSC-Exp, VCS-Exp tries to send it out and then TCP connection fails.
2015-02-09T12:45:04+07:00 DC-VCSE tvcs: UTCTime="2015-02-09 05:45:04,239" Module="network.tcp" Level="ERROR": Src-ip="172.18.7.8" Src-port="25297" Dst-ip="199.19.190.28" Dst-port="5060" Detail="TCP Connection Failed"
2015-02-09T12:45:04+07:00 DC-VCSE tvcs: UTCTime="2015-02-09 05:45:04,240" Module="network.sip" Level="DEBUG": Action="Sent" Local-ip="172.18.8.8" Local-port="7001" Dst-ip="172.18.8.20" Dst-port="25008" Msg-Hash="5130918937689017019"
SIPMSG:
|SIP/2.0 408 Request Timeout

I try to use Free Jabber call to my system via DNS SRV record. It can connect to system but It will disconnect after call for 10-20 sec.
All case which call via DNS SRV record, I checked and found
-The problem happen only call to TPS (via conductor).
-If call to endpoint, It's working fine.
-If call to MCU (registered with VCS), It's working fine.
I see log files which new INVITE is sent by TPS to VSC-Exp, VCS-Exp tries to send it out and then TCP connection fails.
2015-02-09T12:45:04+07:00 DC-VCSE tvcs: UTCTime="2015-02-09 05:45:04,239" Module="network.tcp" Level="ERROR": Src-ip="172.18.7.8" Src-port="25297" Dst-ip="199.19.190.28" Dst-port="5060" Detail="TCP Connection Failed"
2015-02-09T12:45:04+07:00 DC-VCSE tvcs: UTCTime="2015-02-09 05:45:04,240" Module="network.sip" Level="DEBUG": Action="Sent" Local-ip="172.18.8.8" Local-port="7001" Dst-ip="172.18.8.20" Dst-port="25008" Msg-Hash="5130918937689017019"
SIPMSG:
|SIP/2.0 408 Request Timeout

Why does my phone disconnect after 4 hours of call time

why does my phone disconnect after 4 hours of call time? Is there anyway I can change this?

You mean that you're on a single call for 4 hours continously, and then it disconnects?
If so, contact your wireless carrier. That's a cellular connection issue.

Call gets disconnected after playing the menu from CVP

Hello,
When I make a call, the call is going to the ICM and then it is coming to the Call server then to the VXML gateway and then it goes to the VXML server and plays the promt.
The problem i am facing is that once the menu audio is played the call gets disconnected. It doesnt let me choose any options. I feel it is not able to identify the my inputs
I have some debug outputs
Mar 1 01:09:03.095: //-1/C113522D805A/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=1000
   ----- ccCallInfo IE subfields -----
   cisco-ani=sip:[email protected]:5060
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=sip:[email protected];transport=udp
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x67213130, Call Info(
   Calling Number=sip:[email protected]:5060,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=sip:[email protected];transport=udp(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Un
R1#known, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=811, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=65
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
   In: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
   Out: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :cc_get_feature_vsa malloc success
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: cc_get_feature_vsa count is 1
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :FEATURE_VSA attributes are: feature_name:0,feature_time:1732532552,feature_id:23
*Mar 1 01:09:03.103: //65/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))
*Mar 1 01:09:03.111: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
   Event=0x67452800
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/ccCallSetContext:
   Context=0x65097114
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 65 with tag 811 to app "_ManagedAppProcess_bootstrap"
*Mar 1 01:09:03.143: //65/C113522D805A/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Mar 1 01:09:03.147: //65/C113522D805A/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x0
*Mar 1 01:09:03.151: //65/C113522D805A/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
*Mar 1 01:09:03.239: //65/C113522D805A/CCAPI/cc_api_event_indication:
   Event=141, Call Id=65
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=65,
   Caps(Codec=0x0, Fax Rate=0x2, Vad=0x2,
   Modem=0x0, Codec Bytes=20, Signal Type=2)
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=250(ms), Fax Nom=300(ms))
*Mar 1 01:09:04.163: //65/C113522D805A/CCAPI/ccCallHandoff:
   Silent=FALSE, Application=0x6773AB4C, Conference Id=0xFFFFFFFF
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms)
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=0(ms), Initial Digit Timeout=0(ms))
*Mar 1 01:09:04.295: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.299: //65/C113522D805A/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=65
*Mar 1 01:09:04.303: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.307: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:04.311: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms))
*Mar 1 01:09:05.291: //65/C113522D805A/CCAPI/ccAssociateStream:
   Coder=5, DTMF Relay=4, Vad=0,
   Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:05.295: //65/C113522D805A/CCAPI/ccAssociateStream:
   Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:05.327: //65/C113522D805A/CCAPI/cc_api_call_associated:
   Interface=0x0, CallId=65, Disposition=0
   Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
   Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
   TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
*Mar 1 01:09:05.531: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=5000(ms))
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=65
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.539: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:05.543: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms))
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:08.735: //65/C113522D805A/CCAPI/ccAssociateStream:
   Coder=5, DTMF Relay=4, Vad=0,
   Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.739: //65/C113522D805A/CCAPI/ccAssociateStream:
   Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:08.747: //65/C113522D805A/CCAPI/cc_api_call_associated:
   Interface=0x0, CallId=65, Disposition=0
   Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
   Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
   TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
R1#
R1#
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
   Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:16.175: //65/C113522D805A/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Mar 1 01:09:16.179: //65/C113522D805A/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
*Mar 1 01:09:16.227: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
R1#Disposition=0, Interface=0x67213130, Tag=0x0, Call Id=65,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*Mar 1 01:09:16.235: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: :cc_free_feature_vsa freeing 67445940
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: vsacount in free is 0
I have some basic VXML gateway configuration
R1#show run
Building configuration...
Current configuration : 2588 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname R1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 5
ip cef
no ip domain lookup
ip domain name lab.local
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
multilink bundle-name authenticated
voice service voip
sip
header-passing
application
service cvperror flash:cvperror.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service new-call flash:bootstrap.vxml
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service ringtone flash:ringtone.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
service ringback flash:ringtone.tcl
service handoff flash:handoff.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
archive
log config
hidekeys
interface FastEthernet0/0
ip address 192.168.1.253 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
dial-peer voice 9191 voip
service ringtone
session protocol sipv2
incoming called-number 91919191
codec g711ulaw
no vad
dial-peer voice 9292 voip
service cvperror
session protocol sipv2
incoming called-number 92929292
codec g711ulaw
no vad
dial-peer voice 1503 voip
destination-pattern 1503
session protocol sipv2
session target ipv4:192.168.1.57
dtmf-relay rtp-nte h245-alphanumeric h245-signal
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 100 voip
destination-pattern 10..
session target ipv4:192.168.1.100
dial-peer voice 811 voip
service bootstrap
session protocol sipv2
session target sip-server
incoming called-number 81T
codec g711ulaw
no vad
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
end
Please help me out how o fix this issue/

Thanks for the respose
The microapp what I use is user.microapp.ToExtVXML[]
This is the activity log in the CVP application. I dont see any error logs.
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,newcall,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,ani,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,areacode,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,exchange,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,dnis,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,uui,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,iidigits,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.125,CVP Subdialog Start_01,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,Greetings,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.597,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Greetings,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Menu,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.707,Menu,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,Menu,exit,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,how,hangup
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,result,normal
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,duration,1
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,newcall,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,ani,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,areacode,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,exchange,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,dnis,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,uui,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,iidigits,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,CVP Subdialog Start_01,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,Greetings,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.957,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Greetings,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Menu,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,how,app_session_complete
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,result,timeout
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,duration,1819
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,newcall,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,ani,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,areacode,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,exchange,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,dnis,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,uui,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,iidigits,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,CVP Subdialog Start_01,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,Greetings,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.782,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Greetings,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Menu,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,how,app_session_complete
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,result,timeout
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,duration,1829
Thank you

Video call are disconnected after 15 minutes

Hello,
help me pls, after 15 minutes video call disconnected every time.
My configuration CME86 <-SIP-> CME71 <- SIP-> CUCM7.1.3.
I perform call from caller on CME86 to callee on CUCM7.1.3.
I collected debug logs on CME86 and see BYE message from callee on CUCM7.1.3.
Disconnect cause CC/SIP: 16 /200.
Thanks in advance

Cause code 16 is normal call clearing. Does CME71 see the same BYE from CUCM7.1.3? If yes you would need SDI/SDL traces, w/ troubleshooting traces enabled, from CUCM to figure out why it dropped the call.
Please remember to rate helpful responses and identify helpful or correct answers.

Call is getting disconnect after 3 min when the caller is in queue

Hi,
I am using icm 7.0 with ip ivr 7.0. when all the agents are busy and caller is in queue,call is automatically getting disconnected after 3 min with the message " We are experiancing system problem and unable to process the call. please try again later" .
Icm and ivr script are attached
Regards,
Dinesh Joshi

Hi Chrish,
you are right. the timer was for 180 sec only. now things are working fine after increasing the timer.
Thanks a lot for your reply.
Regards,
Dinesh Joshi

Why does my Sennheiser BW 900 headset disconnect after each call?

Although I had no problem pairing a Sennheiser BW 900 headset with my iPhone 4S, it disconnects after each call. It is the only bluetooth device paired to the phone. Any help with solving this problem would be greatly appreciated.

Still having same issue, only now it seems to stay connected at times and disconnect at times. My phone is literally right next to me, I am at my desk, not moving around. The strangest thing is that it REconnects again randomly. So I never know whether to answer with the headphones or using my actual phone! Please help!

My incoming calls on my iphone 5C get disconnected after the first ring. Please help!

My incoming calls on my iphone 5C get disconnected after the first ring. Please help!

I had this problem too with my newly bought iPhone 4s.I went to a technician shop to ask what the problem is and they said they could fix it however they would charge £35 and I believed this was too much. I was playing around with my iPhone to see if I could fix this problem I went on setting the clicked on the music section at the music section there is the button called sound check which was off so I decided to turn it on . After I turned it on my sound came back I was ecstatic I saved myself £35 and having to return the phone.

CME Jabber on iPhone disconnects after 10 Secs

Hi all, have a problem on my jabber clients that they disconnect after 10 seconds when making outbound calls. Router 2921 - c2900-universalk9-mz.SPA.151-4.M4.bin runing CME 8.6. iPhones running Jabber 8.6.5.18781. All other devices (CIPC, 7945's etc. (SCCP)) are working fine and are able to make outgoing and incoming calls.
please assist.
this is what I get from the CME
un 8 15:10:43.459: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Jun 8 15:10:43.459: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Jun 8 15:10:43.459: //1158/E2B9A9F18216/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SPI_EVENT
Jun 8 15:10:43.459: //1159/E2B9A9F18216/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK4291EAA
From: "Dave-iPhone" <sip:[email protected]>;tag=BDEC6C-1325
To: <sip:[email protected]>;tag=gf97ll6v-CC-24
Date: Fri, 08 Jun 2012 15:10:25 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1339168243
CSeq: 103 BYE
Reason: Q.850;cause=102
P-RTP-Stat: PS=539,OS=16170,PR=894,OR=17880,PL=0,JI=0,LA=0,DU=10
Content-Length: 0
Jun 8 15:10:43.495: //1159/E2B9A9F18216/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK4291EAA
From: "Dave-iPhone" <sip:[email protected]>;tag=BDEC6C-1325
To: <sip:[email protected]>;tag=gf97ll6v-CC-24
Call-ID: [email protected]
Timestamp: 1339168243
CSeq: 103 BYE
Content-Length: 0
Jun 8 15:10:43.495: //1159/E2B9A9F18216/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x313A9000
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 7477
Called Number            : 0114674466
Source IP Address (Sig ): 192.168.244.250
Destn SIP Req Addr:Port : 192.168.244.10:5060
Destn SIP Resp Addr:Port : 192.168.244.10:5060
Destination Name         : 192.168.244.10
Jun 8 15:10:43.495: //1159/E2B9A9F18216/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 20
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 192.168.244.250
Source IP Port    (Media): 17620
Destn IP Address (Media): 192.168.244.10
Destn IP Port    (Media): 20040
Orig Destn IP Address:Port (Media): [ - ]:0
Jun 8 15:10:43.495: //1159/E2B9A9F18216/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 102
Disconnect Cause (SIP)   : 200
Jun 8 15:10:43.587: //1158/E2B9A9F18216/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.1:5060;branch=z9hG4bK428F1C;received=192.168.10.1
From: <sip:[email protected]>;tag=BDF344-F6
To: "Dave-iPhone" <sip:[email protected]>;tag=000000000000007f731c4248-4443c002
Call-ID: [email protected]
Date: Fri, 08 Jun 2012 15:18:20 GMT
CSeq: 101 BYE
Server: Cisco-SIPIPCommunicator/9.2.1
Content-Length: 0
Jun 8 15:11:01.471: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
Jun 8 15:11:01.471: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:192.168.244.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK42A3FA
From: <sip:[email protected]>;tag=BE7960-2177
To: <sip:[email protected]>
Date: Fri, 08 Jun 2012 15:11:01 GMT
Call-ID: 38930D6E-B0AF11E1-81BCB8F2-331935B0
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1339168261
CSeq: 11 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Supported: path
Content-Length: 0
Jun 8 15:11:01.487: //1160/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.244.250:5060;branch=z9hG4bK42A3FA
From: <sip:[email protected]>;tag=BE7960-2177
To: <sip:[email protected]>;tag=v7vgdted
Call-ID: 38930D6E-B0AF11E1-81BCB8F2-331935B0
Timestamp: 1339168261
CSeq: 11 REGISTER
Warning: 399 - "SoftX3000 R601-MSG Error ID:[259] LiteFME Tg register ,get Wll param error "
Content-Length: 0
Thanks
David

Just in case you haven't solved this yet, the issue is detailed below.
Workaround
Bind all SIP traffic to a single interface, ideally the voice vlan interface.
   voice service voip
       sip
         bind all source-interface gigabitEthernet 0/0
Cisco Mobile 8.1 integrated w/CME 8.6. On outgoing calls from Cisco mobile the call will connect, get 2 way audio and then disconnect after
15-20 seconds.
The problem is that when the call connects the CME sends a 200 OK to the Cisco Mobile app. The Cisco Mobile app fails to send an ACK to CME, so CME keeps sending 200 OK messages until it times out and ends the call.
The problem is related to CME using a different source IP address for SIP traffic than what was used for the endpoint to register. The IP used as the source address under Voice Register Global is what the TCP connection is set up to for an outgoing call. However the CME uses the best address to source traffic from when it is not bound. As these devices are on the WiFi network (not the wired voice network), the CME uses a different address to source the SIP traffic for these calls than what is defined under the voice register pool. This causes an error in the Cisco Mobile App, which is why it fails to return the ACK.
We bound SIP traffic globally under Voice Service Voip to the same IP that was defined under the voice register global. After this the calls worked fine.
Reference this Junked bug as well:
CSCtu05278   CME Call from Cisco Mobile to IP phone disconnects after
20 seconds

CUCM Call-Disconnect cause 47

Hello all,
i hve an issue with below senario
ph1----CUCM====CME=====NortelCs1000----ph2
1-when ph1 call ph2 , it rings, then silent call, after 12 sec call disconnect
2-debugging CME, gives; CUCM send disconnect cause 47
3-tracing CUCM give
AuConnectErrorInd | waitDisconnect | MediaManager(1,100,141,11) | MediaExchange(1,100,110,11) | (1,100,65,1).12257-(SEP00262D9FC90F:10.0.3.1)| [R:NP - HP: 0, NP: 2, LP: 0, VLP: 0, LZP: 0 DBP: 0]CI1=17696261 CI2=17696262 clearType=0
10.0.3.1is the ip of CME
4-call from ph2 to ph1 give the same symptom with same errors
5-if a phone registered at CME , it can call ph2 or ph1 normally, with codec g729
please help me to resolve this issue as all users cant call each others

Problem is likely a codec mismatch. Do you have MTP checked on the trunk between CUCM and CUCME? When you make calls from the phone to CME, are you sure it is negotiating G729?
47 typically means a transcoder couldn't be invoked when it wanted one. I would start looking at codec to see if you can find out what the issue is.

Remote desktop connects, black screen, disconnects after a few seconds

I have several servers running Windows Server 2012 R2. These servers are used for testing, and are members of a domain which has two DCs, each of which are running DNS Server.
Yesterday I noticed one of the servers was having an issue with its remote desktop connection. If I try remoting into this machine, it connects to a black screen, sits there for 2-3 seconds and then promptly disconnects. No error message is given. I tried
this from a couple different Windows 7 machines, and got the same result. I tried from my wife's iMac, which runs Jump Desktop, and too was met with the same result.
I can, however, remote into this machine via RDP from my work-issued laptop, which is running Windows 8. However, I cannot remote into the problem machine via Remote Desktop Connection from one of the 2012 R2 servers--I am met with the same result as the
other machines: connect, black screen for 2-3 seconds, disconnect without error.
I can remote into all of the other 2012 R2 servers without issue from any machine, including the iMac. I am experiencing this problem on one server only.
Since it is a test server, I reinstalled Windows from scratch this morning. This meant formatting the SSD and wiping everything out. I reinstalled the drivers, joined the server back to the domain and installed all of the updates available from Windows Update.
And rebooted. And yet, the problem persists!
I tried troubleshooting the "usual suspects," like turning off Windows Firewall, disabling antivirus software (currently none installed), disabling IPv6, setting a static IP instead of using DHCP, etc. I did notice the network adapter said it was
on a Network, not a Domain Network. I corrected DNS settings, and the network reidentified itself as being on the ParietalBone.net network. And yet, the problem persists. Remote Desktop to this server still yields a black screen for 2-3 seconds, followed by
a disconnect--EXCEPT from the aforementioned Windows 8 machine.
I looked in the TerminalServices-LocalSessionManager log on the offending server, and there are no errors. I only see repeated entries that say "Session X has been disconnected, reason code 12," where X is 1 or 2. On a couple occasions, the reason
code is 11. But I don't see any errors indicating a breakdown in security negotiation.
Without any robust logs or debug details, I'm at a loss of where to look for a solution.

I am cautiously optimistic that this may have been resolved.
The network card in the computer was only a 10/100 built-in card, so I replaced it with a 1Gbps card. Interestingly, this did NOT resolve the issue. I ran netsh to check on the state of the chimney offload, but it was already disabled.
In the network card properties, I changed the link speed and duplex from automatic to 1Gbps full, re-ran "netsh int tcp set global chimney=disabled" and rebooted. The problem persisted.
And so I went to the network card properties and set IPv4 Checksum Offload, Large Send Offload (three of these -- IPv4, v2 IPv4 and v2IPv6) and TCP Checksum Offload (IPv4 and IPv6) to Disabled. Jumbo Frames were also disabled. I rebooted again.
After the subsequent reboot, I was able to connect to this Server from machines that previously gave me difficulty. The only machine I haven't been able to test today (I'm in the office) is my wife's iMac, which is at home. However, I was able to bring this
server up via Jump Desktop on my iPad, so I'm suspecting that perhaps one of the offload options was to blame. I'm not sure why this would allow me to connect from some machines but not others. And as noted above, I have many other machines running Server
2012 R2 and they are NOT experiencing this problem. And I've not done any tinkering with network settings on any of them, so their chimney and offload settings are all in the default configuration.
Regarding the other items -- NIC team, virtual machine, etc. -- there was no NIC team, no SRIOV and no RDMA involved.
I had to turn IPv6 off on this machine, and only on this machine, because having it enabled was causing a lot more problems. DNS registration would keep failing, it couldn't pull Group Policy, it couldn't get time sync with the DC, etc. The servers are all
set up with static IPv4 addresses. The router supports IPv6 but I'm not actually using it, so disabling IPv6 wasn't a concern to me.

RMI call resets Windows error code?

I'm attempting to retrieve a Windows error code from a remote system, using a simple call to the Windows API function getLastError() which is wrapped inside a JNI function on the remote system. getLastError() returns the last error code set for a particular process/thread.
The problem is, it appears that simply making the remote function call clears the error value before it can be returned. I can confirm that the error value was set on the remote system by printing the Process/Thread ID and getLastError() value to a console window on the remote system when one of my remote methods is called with out-of-range data. If I attempt to retrieve the error code, the Process/Thread ID match but the error code is always 0. No other methods were called on the remote system in between the initial error-generating method and the method which returns the error code.
Note that using the same JNI function calls without RMI works fine - I can generate error codes and retrieve them just fine.
Any ideas? Thanks in advance!

You're assuming that no other system calls are
executed between the one that caused the error you are
interested in and your GetLastError(). RMI executes
system calls on its own behalf, both to return the
value from the previous computation that caused the
error and to dispatch the new call into the method
that does GetLastError(). This approach is never going
to work. Why can't you call GetLAstError() straight
after the error happens, in the same remote method?This is what I suspected is happening, and if true, then I will simply have to rewrite my RMI method calls. Currently, they return -1 on error, and the error number is retrieved with a separate getLastError() call. This is done for legacy reasons (I'm calling C functions which were originally developed without Java/RMI in mind). As suggested, I'll have to call getLastError() immediately after an error occurs, and return the negative value of the error code rather than just -1. I've got about 200 remote methods to modify, so I was trying to avoid this, but hey... it's mostly search/replace.
Thanks for the help.

Calls disconnect after error %CALL_CONTROL-6-NUMBERING_TYPE_NO_MATCH

Similar Messages

Maybe you are looking for