BT's anonymous calls policy
First let me state my opinion on anonymous calls:
Imagine if a heavily disguised person approached your front door, wearing a balaclava, trenchcoat and dark sunglasses, then spoke to you through a voice disguiser, and demanded access to your home, whilst refusing to identify himself.
Would you let him in?
No? Then why would you answer an anonymous call (withheld number)?
AFAIAC anonymous calls are profoundly immoral, totally unjustifiable, and potentially dangerous (scams, stalking, etc.), so why are they not only tolerated, but actually encouraged by BT? I fail to understand why such a sinister thing is even legal in Britain.
So I find it rather perplexing that BT should not only offer this service at all, but actually encourage potential stalkers and scammers with this service for free, whilst incredibly charging the potential victims for the privilege of being protected against this abuse (Anonymous Caller Reject service), at the rate of £16 per quarter plus VAT.
I find this policy completely backwards and morally bankrupt.
Can anyone justify it?
Thanks, but like every other BT customer, I already have Caller Display, since this is now a standard (and free) calling feature on all lines.
It doesn't help though, since the message "Withheld Number" doesn't actually tell me who is withholding that number (obviously), and thus doesn't help me in the slightest to make an informed decision about whether or not to answer the call. Is it a wrong number? Is it a scammer? Is it a friend, who just so happens to misguidedly withhold his number? I have no idea, and absolutely no way to tell.
This is why I use the "Anonymous Caller Reject" service, because it lets the caller know their call has been blocked, and the only way it's going to be accepted is if they allow their number to be seen.
However, this doesn't help much with so-called "Out of Area" calls - calls which are (or pretend to be) from outside BT's network, and whose local network does not forward (or deliberately withholds) CLI (Caller Line Identification) data. Apparently this also includes at least one of BT's own services ("Fixed Text"), which dubiously informs landline owners by automated recording that someone mistakenly tried sending an SMS message to their landline.
Why one of BT's own services is not on their own network (or pretends as much) is quite a mystery. To most people, the endlessly persistent calls from this automated service will be nothing more than a source of curiosity and irritation. To people, like me, who are carpet-bombed by anonymous and "Out of Area" calls, many of which are from criminal groups operating fake lottery syndicates, BT's "Out of Area" calls only further contribute to the terror, despair, and feeling of helplessness. The British police seem just as helpless in the face of these foreign gangs, and quite apathetic with it.
This is just one of the reasons why people need BT's help to protect their privacy and safety. But instead, BT seem to actually be making it easy for their customers to be harassed, or scammed, or worse, to the point where even BT themselves are using "Out of Area" networks (or perhaps just spoofing it) for their own services.
This is what I believe BT and/or Ofcom should be doing to protect British landline (and even mobile) users:
Anonymous calling should be banned outright. End of discussion.
Ditto for fake "Presentation Numbers", which are just as sinister, and worse - deceptive (foreign callers appear to be calling from within the UK)
"Out of Area" calls (real or otherwise) should be classified as anonymous for call screening purposes, and customers should be given the option to block such calls at exchange level. And not have to pay for the privilege.
Meanwhile, Ofcom should give all other networks a deadline by which they must start forwarding accurate CLI data to BT's network, after which time such calls will simply be blocked at the primary/international exchanges. This CLI data must also be verifiable and resolve to the actual line making the call (i.e. not a deceptive "Presentation Number").
The UK government needs to introduce tougher legislation to deal with deceptive and/or clandestine calling practises, against both the perpetrators and those complicit with their behaviour.
Anonymous calls, "Out of Area" calls, and false "Presentation Numbers" are completely unnecessary, unjustifiable, profoundly unethical, sinister, and an actual threat to our safety and privacy, and as such should be banned and criminalised.
Anyone who feels this is an extreme view, is welcome to swap their telephone number with mine, and deal with the consequences.
Similar Messages
-
Block incoming anonymous calls
Hi,
I'm trying to block incoming blocked/unknown/anonymous callers over a sip trunk? I've creaed a translation rule and applied it:
voice translation-rule 5000
rule 1 reject /^$/
voice translation-profile CallBlock5000
translate calling 5000
dial-peer voice xxxx voip
call-block translation-profile incoming CallBlock5000
call-block disconnect-cause incoming invalid-number
To try it out, I'm dialing (from a normal/off network cell and landline) *67 and then the number. This does not work; only if I match the exact number I'm calling from, then it does get blocked.
When I show sip calls during the *67 call I see the calling number is blank.
Calling Number :
When I show sip calls during the regular call, I see the proper Calling Number.
As I understand it, with Call Manager and phones running SCCP, I cannot enable/use anonymous call blocking; so I do have to enforce the call blocking policy at this gateway device (UC520).
I'm very new to Cisco voice, so sorry I'f I'm missing something obvious. Thanks in advance!Thanks, I tried the rule--still private gets through.
*Feb 14 21:52:37.457: //394/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 207.2.123.180:5060;branch=z9hG4bK71b5401f596084541bb0894ae16bbbc8
From: ;tag=3538245046-512161
To: [email protected](CUBE external IP)>
Date: Tue, 14 Feb 2012 21:52:37 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 14 21:52:37.457: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected](UCMBE IP):5060 SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP)>;:5060;branch=z9hG4bK2B4F4E
Remote-Party-ID: [email protected](UC500/CUBE IP)>;party=calling;screen=no;privacy=full
From: "anonymous" ;tag=4C78DEC-1E4B
To: [email protected]>
Date: Tue, 14 Feb 2012 21:52:37 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0186175275-1452085729-2206375728-3952970709
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1329256357
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 2063 472 IN IP4 yy.yy.yy.yy(UC500/CUBE IP)
s=SIP Call
c=IN IP4 yy.yy.yy.yy(UC500/CUBE IP)
t=0 0
m=audio 19566 RTP/AVP 0 101
c=IN IP4 192.168.20.5
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Feb 14 21:52:37.465: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Date: Tue, 14 Feb 2012 21:50:47 GMT
From: "anonymous" ;tag=4C78DEC-1E4B
Allow-Events: presence
Content-Length: 0
To: [email protected](UCMBE IP)>
Call-ID: [email protected]
Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP):5060;branch=z9hG4bK2B4F4E
CSeq: 101 INVITE
*Feb 14 21:52:37.469: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Date: Tue, 14 Feb 2012 21:50:47 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "anonymous" ;tag=4C78DEC-1E4B -
SIP Invite change for anonymous calls
I noticed a change in IOS Gateways in how it deals with anonymous calls. Anonymous calls in version 12.4(25g) generates an SIP INVITE:
From: "anonymous" <sip:[email protected]>
A anonymous calls on version 15.1(4)M8 generates a SIP INVITE:
From: "anonymous" <sip:[email protected]>
The p-asserted-identity and remote-party-id did not change. None of our other SIP systems use the "anonymous@" format.
How do I get the 15.1 GW to use "<number>@" instead of "anonymous@" for these calls?
Thanks,
-JohnHi John,
The only possible solution that i could think of for this scenario is through the use of SIP profiles on the gateway. There are quite a few posts and docs which you can check to try and configure one for your setup
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/105624-cube-sip-normalization.html
http://www.gossamer-threads.com/lists/cisco/voip/127376
HTH
Manish -
Anonymous calls, Unknown Callers and 10 number limit on call blocking feature
I have FIOS Digital Voice. I hope that someone can give me information to help me out - because like many others here, right now I am regretting having this and have it unplugged more than plugged in - for a service I am paying for.
My number is unlisted.
In my account I have it checked to reject anonymous calls. However, I am still getting calls that are "unknown callers" and "unavailable" numbers. So, if I have it set to reject these calls why am I still receiving them.
Also, why is there such a low limit of adding blocked phone numbers to your account? It is almost like a part-time weekly job to rotate through all the numbers adding new ones, deleting old ones - and they are able to do this much easier, quicker, and they are getting paid, while I am the one paying for this service - which isn't working.
Can anyone let me know why the unknown callers are still getting through and if there is a way to add more than 10 numbers to the blocked list?
As it is, my home phone is unplugged because it has become a useless payment unless this can be addressed.
Thanks
BTW - Are the web/tech guys aware that when you are signed into your account, if you click on "support" from the top menu for "phone", it takes you to a completely different area and asks you to sign in again - but if you click on the "support from the left menu for "phone" it takes you where you need to go. But the website is a whole other can of worms.This might help answer your questions.
http://www.dslreports.com/forum/r27058664-PHONE-Anonymous-Call-Rejection-and-qUnavailable-Unknown-N -
Regular Expressions with Call Policy on VCSe
Hi Guys,
I am working on firming up the call policy on my VCS Expressway to try to better intercept the SIP spam requests it is getting from internet ip numbers. Right now those spam requets are getting rejected by the loop detection but I want to intercept them before they even do a search on the Expressway. It seems that the call policy rules I create without regular expressions are functioning fine but I don't think I have the syntax correct for the regular expressions.
The goal of this rule is to reject any incoming SIP request that has a destination alias format of 7 to 17 digits followed by an @VCSe_IP. so for example it would reject the following attempts: 0123456@VCSe_IP and 0123456789101112@VCSe_IP with one rule.
The policy I created is this: source pattern: unauthenticated user, Destination pattern: \d{7,17}@xx\.xx\.xx\.xx (where xx is the individual octets of the VCSe IP address), Action: reject
However the above policy does not seem to be rejecting the calls before they do a search. I have checked the above expression with the check pattern tool on the VCSe and it comes up with a sucessful match when I try the destination alias of a request that made it through, hence my confusion. Any help you guys could provide would be appreciated.
Thanks,
StevenSteven,
Default Zone access rules do not relate to this at all and you can keep those set to 'No'.
How exactly are you placing the test calls when attempting to verify this?
I created the following CPL rule on my X7 VCS (With 10.10.10.10 being the IP address of my VCS):
Source pattern:
Destination pattern: \d{7,17}@10\.10\.10\.10
Action: Reject
I then proceeded with placing a SIP call from an unregistered C20, calling the URI '[email protected]' while running a diagnostics log on my VCS with Network log level set to 'DEBUG', and captured the following in that log:
Incoming INVITE:
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,598" Module="network.sip" Level="INFO": Src-ip="10.x.x.x" Src-port="5060" Detail="Receive Request Method=INVITE, Request-URI=sip:[email protected], Call-ID=9dd19ad75b1063ecf716461b149e9e2a"
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,598" Module="network.sip" Level="DEBUG": Src-ip="10.x.x.x" Src-port="5060"
SIPMSG:
|INVITE sip:[email protected] SIP/2.0
Call processing logic, showing CPL matching:
2013-02-22T16:03:36+01:00 vcs02 tvcs: Event="Search Attempted" Service="SIP" Src-alias-type="SIP" Src-alias="10.x.x.x" Dst-alias-type="SIP" Dst-alias="sip:[email protected]" Call-serial-number="0886391c-7d01-11e2-adf5-0050569a08fd" Tag="08863aac-7d01-11e2-bd2e-0050569a08fd" Detail="searchtype:INVITE" Level="1" UTCTime="2013-02-22 15:03:36,601"
2013-02-22T16:03:36+01:00 vcs02 tvcs: Event="Call Attempted" Service="SIP" Src-ip="10.x.x.x" Src-port="5060" Src-alias-type="SIP" Src-alias="sip:10.x.x.x" Dst-alias-type="SIP" Dst-alias="sip:[email protected]" Call-serial-number="0886391c-7d01-11e2-adf5-0050569a08fd" Tag="08863aac-7d01-11e2-bd2e-0050569a08fd" Protocol="UDP" Auth="NO" Level="1" UTCTime="2013-02-22 15:03:36,601"
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,602" Module="network.cpl" Level="DEBUG": Remote-ip="10.x.x.x" Remote-port="5060" Detail="CPL: "
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,602" Module="network.cpl" Level="DEBUG": Remote-ip="10.x.x.x" Remote-port="5060" Detail="CPL: "
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,602" Module="network.cpl" Level="DEBUG": Remote-ip="10.x.x.x" Remote-port="5060" Detail="CPL: matched "
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,602" Module="network.cpl" Level="DEBUG": Remote-ip="10.x.x.x" Remote-port="5060" Detail="CPL: "
VCS responds to INVITE with 403 Forbidden:
2013-02-22T16:03:36+01:00 vcs02 tvcs: UTCTime="2013-02-22 15:03:36,616" Module="network.sip" Level="DEBUG": Dst-ip="10.x.x.x" Dst-port="5060"
SIPMSG:
|SIP/2.0 403 Forbidden
As you can see, on my VCS everything seems to work as expected. I'd recommend you capture a similar diagnostics log on your own VCS to check what is different in your test call compared to the output above. -
Anonymous Call block not working
I had the 'Anonymous call block' feature included with my bundle but I am still getting calls from anonymous callers, telemarketers of course, on a daily basis.
Help?Hello mhicks,
In your original post you only said "Help?"
You received some things that should help. at the top of all the folders is a disclaimer "These Forums are a peer-to-peer support channel in which users help other users. If you need an answer from Verizon directly please Contact Us. ".
If you are here wanting someone at Verizon to answer "how my order was never filled correctly to begin with and that I am in fact paying for services not rendered." then you are unlikely to ever find an answer here.
People here are good at helping others solve problems and a few Verizon staffers chime in with solutions as well but the specific question you are looking for an answer to won't likely get answered.
Hope you get things working the way you want. I hate telemarketers as well. Fortunately the donotcall list has been working pretty well for me. -
How do I block "PRIVATE CALLER" or "ANONYMOUS" calls?
I am receiving SCAM calls that show PRIVATE CALLER or ANONYMOUS on the CALLER ID
How do I BLOCK these calls?Check out this link:
http://www.verizon.com/support/residential/phone/homephone/calling+features/anonymous+call+rejection...
If you have FiOS Digital Voice, you can control the feature from the FDV web page.
Enjoy.
If a forum member gives an answer you like, give them the Kudos they deserve. If a member gives you the answer to your question, mark the answer as Accepted Solution so others can see the solution to the problem. -
Can I configure an SPA112 to forward anonymous calls?
Can I configure an SPA112 to forward anonymous calls?
There is a call forward facility, but how to a match a withheld CLI?Can I configure an SPA112 to forward anonymous calls?
There is a call forward facility, but how to a match a withheld CLI? -
Do I have to turn on Anonymous Call Rejection
Our Verizon service says that we have Anonymous Call Rejection. However, we seem to get a lot of anonymous calls. Do I have to turn the Anon. Call Rejection on?
Thanks.
Solved!
Go to Solution.I found the answer by searching this forum. Should have checked first!
-
Call Blocking on Anonymous Calls
What prefix code can I use to allow me to call a number that is blocking anonymous calls. I am ex-directory. This is one for Devon-David I think.
Solved!
Go to Solution.Hi and welcome.
Not sure I fully understand the question.
If you want to call a number and you dont want them to see your number you would use the prefix 141.
If your own number is currently witheld and by you calling someone you want them to see your number the prefix is 1470.
Hope that helps, if not perhaps just run me through the scenario of what you want to acheive.
(If I have helped you in any way to say "Thank You" please click on the star next to the message. Thank You)
If I have solved your Issue please click the "Mark as accepted solution" button. -
Anonymous Call Rejection not working
I have anonymous call rejection on my phone, but it doesn't seem to work. Several times a day I get a call that shows up as "Not_Available" "Unknown Number."
Is anyone else experiencing this?YES, the past few months I have experience many more calls from tele-marketers, timeshare resellers and small business vendors offering their services, Although I am registered with the National Do Not Call Registry, they still try to call me at my residential phone all hours of the day.
I depend upon my home phone provider to offer the feature of 'anonymous call rejection' and have entered the *77 to block them
but
this feature is not working as advertised.
Enhancement request, stop sending through calls where CallerID 'Not_Available'. I have no way of knowing whom is calling, nor do I have a way to report a complaint since I cannot identify the company or phone number calling me. Cannot see a call log of incoming calls either. -
Problem with anonymous caller id in spa8000
Hi,
We have started to migrate our customer from pap2t to spa8000.
Now we got an issue that we need help with.
If we activating CID block (*67) the sip server will deny the INVITE
I think it has to do with from and contact field.
All parts is now anonymous, in pap2t the real number is still in the <>.
Is it possible to change this?
Any Ideas?
My invite looks as following:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5360;branch=z9hG4bK-ff7fa670
From: "Anonymous" <sip:anonymous@localhost>;tag=1fdbc659fb942988o0
To: <sip:[email protected]>
Remote-Party-Id: "33333333333" <sip:[email protected]>;screen=yes;privacy=full;party=calling
Call-Id: 612a4f6d-18571384@localhost
Cseq: 101 INVITE
Max-Forwards: 70
Contact: "Anonymous" <sip:[email protected]:5360>
Expires: 240
User-Agent: Linksys/SPA8000-6.1.12
Allow-Events: talk, hold, conference
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
Date: Mon, 02 Dec 2013 15:29:48 GMT
with pap2t is look like this:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 3.3.3.3:5061;branch=z9hG4bK-1ff53e0
From: Anonymous <sip:[email protected]>;tag=c0bc166c356c37ffo1
To: <sip:[email protected]>
Call-Id: 8d9cc4b0-68e7be8b@localhost
Cseq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous <sip:[email protected]:5061>
Expires: 240
User-Agent: Linksys/PAP2T-5.1.6(LS)
Content-Length: 438
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
Date: Mon, 02 Dec 2013 19:01:54 GMT
//NJust to be sure - do you know you are replacing one end-of-life platform (PAP2T) with another end-of-life device (SPA8000) ?
But back your question.
Your's PAP2T's INVITE has no CLIR active. Caller number is present in both From: and Contact: header, there is no Privacy: Id nor Remote-Party-Id.
So, next-step exchanges know nothing about CLIR and caller's number is transferred to called phone with no restrictions.
On oposite side, SPA8000's invite claim CLIR using Remote-Party-Id method. But your's SIP gateway is rejecting such call.
You either have no right to use CLIR or particular SIP gateway is configured to use other way to request CLIR Remote-Party-Id. Or the CLIR CALL request is rejected somewhere on the path to destination.
Simplest solution ? It seems that CLIR did not worked on your's former network correctly and caller's number has been delivered to called user with no restriction. So disable CLIR service on SPA8000. No functionality will be lost.
Message was edited by: Dan Lukes (SPA8000 is not EOL) -
Unknown /Private /Anonymous call blocking for specific extensions - CUCM 7.5.3
Hello ,
Can anyone explain is there anyways to block only anonymous/Private/Unknown calls to specific DN's in CUCM 7.5.3?
Regards,
RamThanks for your revert....
Sorry ,Yes you are right we are using CUCM 7.1.3.32900-4 and we are not using H.323 gateway.
In that case ,I will look for system to be upgraded to 8.0 or later version to make it work as stated by brianmeade
Thanks,
Ram -
CUCM 10.5 + MGCP VG Blocking 'Anonymous' calls
Hello to all.
I currently have a CUCM 10.5 installation with a 2921 VG configured with MGCP.
The problem experienced is that when someone calls with hidden caller id, the provider sends the Calling Party Number as 'anonymous'.
This causes CUCM not to process the call and reject it.
The odd thing is that I have another customer with CUCM 5.X with a similar setup (MGCP VG & anonymous as the caller ID ) where calls are processed without a problem.
I've searched the forums and saw other experincing the same issue as we are, even with CUCM 7+, however no answer or solution was provided.
Changing to H323 is not an option at this point.
Any input would be highly appreciated.
Many thanks,
Chris.Hello Carlo,
Here is the output from the VG when dialing inwards through a phone with CallerID disabled.
I also just spoke with Cisco TAC and they have told me there is no current work-around and that we need to change the VG to H323.
Please let me know if you have any suggestions.
Thanks,
Chris.
Nov 4 11:47:21.977: ISDN Se0/1/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x9300
Date/Time i = 0x0E0B040D2F15
Date (dd-mm-yr) = 14-11-04
Time (hr:mnt:sec) = 13:47:21
Nov 4 11:47:21.985: ISDN Se0/1/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x1300
VG2921#
Nov 4 11:47:24.413: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x15DC
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Calling Party Number i = 0x0180, 'anonymous'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '4321586251'
Plan:ISDN, Type:Unknown
Nov 4 11:47:24.417: MGCP Packet received from 192.168.3.10:2427--->
CRCX 52335 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c5e000000F5800015dc
X: 1
L: p:20, a:PCMU, s:off, t:b8
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
VG2921#
Nov 4 11:47:24.421: MGCP Packet sent to 192.168.3.10:2427--->
200 52335 OK
I: 28B9
v=0
c=IN IP4 192.168.3.9
m=audio 21444 RTP/AVP 0 100
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
<---
Nov 4 11:47:24.425: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x95DC
Channel ID i = 0xA98381
Exclusive, Channel 1
Nov 4 11:47:24.425: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x95DC
Cause i = 0x8095 - Call rejected
Nov 4 11:47:24.449: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x15DC
Nov 4 11:47:24.449: MGCP Packet received from 192.168.3.10:2427--->
VG2921#DLCX 52336 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c5e000000F5800015dc
I: 28B9
X: 1
S:
<---
Nov 4 11:47:24.469: MGCP Packet sent to 192.168.3.10:2427--->
250 52336 OK
P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=0, LA=0
<---
Nov 4 11:47:24.469: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x95DC
VG2921#
Nov 4 11:47:40.401: MGCP Packet sent to 192.168.3.10:2427--->
NTFY 902475903 *@VG2921.cucm.local MGCP 0.1
X: 0
O:
<---
Nov 4 11:47:40.401: MGCP Packet received from 192.168.3.10:2427--->
200 902475903
<---
VG2921#
Nov 4 11:47:48.249: MGCP Packet received from 192.168.3.10:2427--->
MDCX 52337 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c4a000000F5000012fb
I: 28B3
X: 1e
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
Nov 4 11:47:48.249: MGCP Packet sent to 192.168.3.10:2427--->
200 52337 OK
<---
Nov 4 11:47:48.253: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x12FB
Cause i = 0x8090 - Normal call clearing
Nov 4 11:47:48.289: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x92FB
Cause i = 0x8190 - Normal call clearing
Nov 4 11:47:48.289: MGCP Packet received from 192.168.3.10:2427--->
VG2921#DLCX 52338 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c4a000000F5000012fb
I: 28B3
X: 1e
S:
<---
Nov 4 11:47:48.305: MGCP Packet sent to 192.168.3.10:2427--->
250 52338 OK
P: PS=5510, OS=881600, PR=5497, OR=879520, PL=0, JI=0, LA=0
<---
Nov 4 11:47:48.309: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x12FB
VG2921#
Nov 4 11:47:54.449: MGCP Packet sent to 192.168.3.10:2427--->
NTFY 902475904 *@VG2921.cucm.local MGCP 0.1
X: 0
O:
<---
Nov 4 11:47:54.449: MGCP Packet received from 192.168.3.10:2427--->
200 902475904
<---
VG2921# -
Missed CAll from Anonymous CAller
There is a user gets the below message when the calls come from external
what is the solution
Mashhour Farajtoo many things could cause that
Number manipulation on the Trunk or Gateway
Number is hidden by the PSTN network itself which mean they caller has this feature for his/her number
calling from a phone box in the street
and so on....
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Mostafa Eltohamy
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