Configure toll fraude prevention c880voice

Hello: I need to configure toll fraude prevention in three of my routers, Ios versions:
c880voice-universalk9-mz.124-24.T6.bin
c880voice-universalk9-mz.150-1.M8.bin
c880voice-universalk9-mz.150-1.M8.bin
I used the following guide, which according to Cisco, is intended for ios version 12.1 or later:
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/107626-cme-toll-fraud.html
This is the configuration I planned to use, but my routers do not support any of the commands under "voice service voip",
Router> enable
Router# configure terminal
Router(config)# voice service voip
Router(conf-voi-serv)# ip address trusted authenticate
Router(conf-voi-serv)#ip address trusted call-block cause call-reject
Router(conf-voi-serv)# ip address trusted list
Router(cfg-iptrust-list)#ipv4 10.0.0.0 255.0.0.0
Router(cfg-iptrust-list)#end
I have been seaching for equivalent commands but haven't found any.
Is there anyone with experience configuring toll fraud prevention for devices with IOS release older than 15.1(2). What am I doing wrong?
Thanks in advance!

As Ayodeji suggested, it is recommended to upgrade the IOS for this feature.
Or esle, you can try configuring Access-Lists to block the traffic coming from un-wanted IPs

Similar Messages

Disable transfer to international number in VoiceMail on BE3000 to avoid toll-fraud

Is there a way that we can disable “transfer to international number feature” in Voicemail setting to prevent toll fraud on BE3000 ? I can't find this setting in BE3000 administration GUI and there is no Cisco Unity Connection GUI that we can login to modify this feature

Hi,
I am not sure which setting are you referring to? Please provide screen capture.
Thanks

Toll fraud traffic pumping on a Call manager express and unity express with an 800 toll free number.

I have a customer with a call manager express and a unity express. They have an 800 toll free number that rings in to an auto attendant on the CUE. The problem is something called toll fraud traffic pumping. The caller, always from a different number, reaches the auto attendant by dialing the 800 number. They then somehow manage to keep the line connected for up to 12 hours by somehow staying in the auto attendant. I opened a TAC case and was informed that it is not in Cisco's hands to create a way to disconnect a call that is connected to the CUE. I need a way to have a call that stays in the auto attendant or voicemail after 3 to 5 minutes. I can't see a way to do it in the auto attendant script. Can someone help with a method that will disconnect the caller if they stay in the CUE for a period of time?

Thank you for the response Jaime.
That is what TAC told me. I am not familiar with TCL scripting and I have been looking at some guides on it. Any advice on a source to go to for assistance with that, possibly someone to help?

Verizon Fraud Prevention Team Number

Is the Verizon Fraud Prevention Team number not working today, or are they just really busy? I've been trying to call all day to verify my order, but haven't been able to get through. Thanks.

Figures, right after I posted this, after being on hold for a while, someone answered and my order was verified.
Thanks

Is there a configuration option to prevent an unprivileged user from accessing the firefox profile manager and/or firefox safe mode?

I'm designing a locked-down Firefox user profile for use on public computers (common room in an apartment building). I can use existing plugins and add-ons to prevent access to about:config and to lock down the various firefox preferences but this is moot if a user can still access the firefox profile manager or can start firefox in safe mode. Is there any configuration setting that could prevent this?

Hi...
Reinstalled 10.7.3 from the Combo Updater from apples website.
The only way to reinstall the Mac OS X or repair the startup disk running v10.7.3 Lion, is to use Lion Recovery The combo update does not do that.
How much free space on the startup disk? Not enough free space can account for the problems with your apps.
Right or control click the MacintoshHD icon. Click Get Info. In the Get Info window you will see Capacity and Available. Make sure there's a minimum of 15% free disk space.
and no web-pages will load.
Try using OpenDNS as suggested here > Safari 5.0.1 or later: Slow or partial webpage loading, or webpage cannot be found
Use OpenDNS for better speed, more security, includes anti phishing filters, prevents browser redirects, and it's free.
Open System Preferences / Preferences then select the Network tab. Click the Advanced tab then click the DNS tab.
Click +
Enter these addresses exactly as you see them here.
208.67.222.222
Click +
208.67.220.220
Then click OK.
edited by: cs

Offnet to Offnet call fraud prevention

Hi all,
I was able to block offnet to offnet call transfers in my CUCM 7. environment thanks to the "Block OffNet to OffNet Transfer Service Parameter". But I have discovered one scénario that allows call fraud though:
We have to internal callers A and B and the External Callers C and D.
C calls A (the call is marked as offnet by the gateway so A won't be able to transfer to D for exemple)
A transfers to B (the call looses its offnet classification and becomes onnet)
B transfers to D
Is there any configuration that can be done to block this scénario
Best Regards,

Dear All,
Then, this seems to be a loophole in cucm. I was eagerly trying to find out something which blocks it or do trick, so firstly I came with below service parameter. I am not sure this really will do something here or not because I dont have a cucm in lab for testing so can't test it out.
Can please give your comments on it, will it effect ?
Use Original Call Classification for Transferred Calls:
Description: This parameter determines whether Unified CM classifies a call as internal or external during and after a call transfer based on the classification of the original call that was transferred. Call classification affects call treatment such as ring tone, forward destinations, and caller ID display. Valid values specify True (Unified CM classifies the call based on the original call classification) or False (Unified CM classifies the call as internal because transfers originate from an IP phone which is always internal). For example, when this parameter is set to True and an external (OffNet) call is received and transferred by an internal phone, Unified CM classifies the transferred call as external even during the consultation call. The ring tone that is used for the consultation call is the external ring tone and during the consultation call (including blind transfers), the selection in the Ignore Presentation Indicators (internal calls only) checkbox on the transfer destination's Phone Configuration window ceases to apply because the call is now considered external.
This is a required field.
Default: False
Suresh

IOS XE Anti toll-fraud

Hi Guys,
I see that I have following commands on CUBE Ent on an ASR running IOS XE
voice service voip
ip address trusted list
ipv4 x.x.x.x 255.255.255.255
ipv4 y.y.y.y 255.255.255.255
But at the same time can find that is supported on IOS 15.1 and up series only on cisco website.
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080b3e123.shtml
Can I be sure that this also corresponds to an ASR IOS XE, since I have those commands?
Thx!

If you have them, you can use them.

Toll Fraud Protection - How to enter /24 range?

A VOIP provider uses a 74.122.242.0/24 network range for incoming and outgoing VOIP services.
Their setup documents request that we allow the entire range.
Typical of Cisco, the CCA only allows individual IP addresses to be entered.
Can someone inform me of a way to add the entire /24 without having to enter each address into the CCA individually?

Hello Brook,
I am not sure if it could be added through CCA maybe CLI only.
Best regards,
Alex

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
   Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source           IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone

Problem with H323 gateway

We have a CUCM ver 8.6.2.22900-9.
For our PSTN access we have the following scheme:
CUCM <========> Cisco2911 <======> ISP-------> PSTN
H323 tk SIP tk
Our Cisco2911 is running ver 150-1.M4.
We configure Cisco 2911 as H323 gateway on CUCM, and we established a SIP tk with our ISP.
When we tested incoming calls we received them at router but they are unable to acceess CUCM,
When we tested outgoing calls we got reorder tone and they did not reach our voice gateway.
We run "debug cch323 all" and tested with one incoming and one outgoing call, and we got the out on attached file "debug h323 forum"
CUCM Ip : 10.1.2.2
Router's H323 int: 10.2.2.1
Do you have any idea about the root cause?
Can be an access list problem preventing proper H323 traffic??
Thanks in advanced
Enrique

Do you have voip dial-peers pointing back to your cucm from your gateway? If not then the call wouldn't route onward to the cucm, but the gateway would also ignore communications from that souce because of toll fraud prevention.
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/112083-tollfraud-ios.html

FXO not giving the dial tone

Hi,
We have FXO ports on the router and SIP trunk towards ITSP. People used to dial into FXO and get the dial tone to callout using the SIP trunk. But after the upgrade of the IOS, this functionality has stopped. Now if you call the FXO, you get busy tone. The IOS has been upgraded from 12.4(11) to 15.1(M4) to basically its a big leap.
I strongly believe that we need to make some configuration to make it work like before.
Please advise.
Attachached are the logs from the "debug vpm all"

[+] for Calro
Here is the new feature complied in 15.X release
http://www.cisco.com/en/US/docs/ios/15_1/release/notes/151-2TNEWF.html
Toll Fraud Prevention
In Cisco IOS Release 15.1(2)T, the Toll Fraud Prevention feature is supported as below:
•Source IP address authentication is enabled on incoming IPv4 H323/ or SIP trunk calls. The source IP address of any incoming IPv4 H323 or SIP trunk calls will be authenticated based on:
–Manually configured IP address trusted list.
–VoIP dial-peer session target (the state of a VoIP dial-peer must be in "Operation State = UP")
Incoming IPv4 H323 or SIP trunk calls will be rejected if the authentication fails and the default cause-code call-reject (21) disconnects the call.
Execute the show ip address trusted list command to display IP address trusted data and a list of valid source IP addresses. The default behavior can be disabled as shown in the example below:
voice service voip
no ip address trusted authenticate
•Secondary dial-tone is disabled for a call initiated from a FXO port. No secondary dial-tone causes the outgoing call setup to fail if the called number is NULL. The default behavior can be disabled as shown below:
voice-port
secondary dialtone
•Direct-inward-dial is enabled to prevent the toll fraud for incoming ISDN calls. Two-stage dialing is disabled for incoming ISDN calls by default. The incoming called number will then be used for outgoing call setup. The default behavior can be disabled as shown in the example below:
voice service pots
no direct-inward-dial isdn
For more information, see the Cisco Unified Communications Manager Express System Administrator Guide at the following URL:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm.
html
Br,
Nadeem
Please rate all useful post.

IOS 15.1 POTS dial-peer problem

Good day!
After updating to IOS version 15.1(4)M6 there is a problem with outgoing calls through the PRI.
Router configuration in the attach. On the gateway turned on debugs:
sho debugging
The following ISDN debugs are enabled on all DSLs:debug isdn error is ON.
debug isdn q931 is ON. (filter is OFF)
DIALPEER:
debug voip dialpeer error call is ON (filter is OFF)
debug voip dialpeer error informational call is ON (filter is OFF)
debug voip dialpeer error software is ON
debug voip dialpeer error software is ON informational
debug voip dialpeer function is ON (filter is OFF)
debug voip dialpeer inout is ON (filter is OFF)
debug voip dialpeer detail is ON (filter is OFF).
The result of debug also in attach.
On CUCM 9.1 configured SIP trunk to this gateway. When I call the test number 89261234567 I hear a busy signal. If I upload this version 12.2(24)T6, with exactly the same configuration, everything works successfully. Accordingly some problem lies precisely in the new IOS. Please tell me to what may be the case.
Thank you!

Hi Filipp,
As Deji correctly guessed, the call is disconnected because of the Toll Fraud prevention.
*Feb 4 15:35:26 MSK: //1/6FAB58800001/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 1 with tag 101 to app "_ManagedAppProcess_TOLLFRAUD_APP"
*Feb 4 15:35:26 MSK: //1/6FAB58800001/CCAPI/ccCallDisconnect:
Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
you can add the IP addresses of CCM servers from where the calls are originated under
voice service voip
ip address trusted list
URL for your reference: https://supportforums.cisco.com/docs/DOC-12228
Please rate all the useful posts

Router 2911 with voice card VIC3-E/M VOIP Problem

hi All
I want to ask if any of you have any experience like this. I have deployed a new Cisco router (model 2911) in my client. The previous router they used is 2811. After deployement, the WAN & LAN connection work fine, but the VOIP connection does not work.
The previous router (2811) uses a voice card module VIC2-e/m. But the new router (2911) use voice card module VIC3-e/m. It is used to connect to the PABX Panasonic TD-1232.
Could this be a factor?
Can anyone help?

Hi,
In addition to what Terry has said, the only change is IOS 15.1 where Toll Fraud Prevention feature was introduced which says
The router will automatically add any destinations that are defined as an ipv4 target in a VoIP dial-peer to the trusted source list.You can run the command
show ip address trusted list
To overcome this behaviuor, you can follow the three ways as suggested in below link .The one has been already suggested by Terry.
https://supportforums.cisco.com/document/46566/understanding-toll-fraud-enhancements-1512t
your 28XX router has 12.4 IOS version
regds,
aman

Preventing Skype Users Accessing a Toll Free Numbe...

Is it possible to request from Skype to block Skype users from dialing a specific Toll Free Number?
I work for a Teleconferencing Company and it seems like Fraudsters are using Skype to dial a Toll Free Number of ours and continually attempt to commit toll fraud on our bridges.
Because it is free to call Toll Free numbers it is allowing these people to make infinite amount of calls to try and hack our system.
So the alternate question, are Skype callers identified to telecommunications carriers and therefore can I have my Telco block Skype calls to my toll free numbers?

Hi, David_ray, and welcome to the Community,
"Toll Free" numbers in Australia may not be included in your subscription; Skype Customer Service can clarify this.
Here is a link to the instruction on how to contact Skype Customer Service via their secure portal: Contact Customer Service
Here are a few more notes to guide you along the way ...
You will need to proceed through each step, one at a time. Sign on to the Skype website as requested. Choose the subject and topic which most closely matches the item you need assistance with. Then, continue past Step 2 of the instruction where several articles from the FAQ library will appear for you to review, and proceed to Step 3, Continue Support Request (the blue "button" appears at the lower right corner of the website page). You may also skip through Step 4 where you will be referred back here to the Community; no need to do this as the Community is where you started. When you complete the webform and click Submit, your information is relayed to Skype and you will receive a reply via e-mail.
If you are a Skype Premium or other eligible customer, you will be routed to start an instant message chat session with a Customer Service agent.
Regards,
Elaine
Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
Trustworthy information: Brian Krebs: 3 Basic Rules for Online Safety and Consumer Reports: Guide to Internet Security Online Safety Tip: Change your passwords often!

Need configuration help on producing dial tone

Hello Experts,
I have a Cisco 2921 router with VWIC3-2MFT-T1/E1 card. On this card we have T1-CAS digital line connected. We have been provided with a set of DID numbers. We have a requirement where, when we dial a DID, the router should provide a dial tone, and should allow the user to dial to extension numbers. Not sure if this is feasible. If at all possible, will need to some configuration help.
Thanks
Arabinda

Sure it's possible. What's the T1 connected to? The router will offer two-stage dialing (aka dial tone) when the incoming POTS dial-peer does not have the 'direct-inward-dial' command on it. The router will accept any input and search for an outbound dial-peer (or ephone-dn for locally registered DNs) to match. Be careful if the T1 is connected to the PSTN as this is a toll fraud risk. You need to use CoR to reign in what outbound dial-peers are available to it.
Dial Peer Basics:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Class of Restrictions:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a008019d649.shtml
Please remember to rate helpful responses and identify helpful or correct answers.

Configure toll fraude prevention c880voice

Similar Messages

Maybe you are looking for