Restrict Caller ID Line from SIP

[Louis-Philippe Descamps]

Hi

We are receiving external calls from 000000 and i want to hide 000000 and show display.

I setup a TP with pattern 000000 but i cant hide the 000000. I know that cucm is using that TP because i can block/route the call.

i can put 1234 as Calling Party Transform Mask and i will see 1234 but cant find a way to make it "unknown"

any idea?

thanks

[Ayodeji Okanlawon]

Louis,

Try and change the TP "
Calling Line ID Presentation" to restricted

[Louis-Philippe Descamps]

Hi Ayodeji,

it's already set to restricted. thanks

[Deepak Mehta]

One way of doing it is which i have tested is...

On SIP Trunk inbound CSS - match a Translation pattern for called number and then restrict the Calling line id presentation before sending the call to the actual DN.Thanks

[Louis-Philippe Descamps]

hi

it can be any called number.... i need to do it on the calling TP. see my diagram, if cli = restricted etc... it will display 000000, i need to change that to restricted. my user DN are in eu_internal partition.

thanks

[Deepak Mehta]

Thanks ,it is clear now what you are trying to achieve .Just trying to  think loud here.

Can you try restricting it on SIP trunk level and reset the trunk , i know this is not what you are looking for however this will give us an idea if calling restriction works or not in this set up.

Or instead of unknown you can try * or + any other supported character.

[Louis-Philippe Descamps]

hi

i restricted it on the trunk and reset but still the same.

i put a + and it displays plus

it used to work

thanks

[Ayodeji Okanlawon]

So what does the display on the phone show when you set calling line ID to restricted?

[Louis-Philippe Descamps]

shows the caller cli

[Ayodeji Okanlawon]

We will need to look at CUCM logs to see what is going on. I just tested this now on my translation pattern from a call that came in via a sip trunk and it showed "private" Can you send us CUCM logs..

[Louis-Philippe Descamps]

attached

thanks

[Ayodeji Okanlawon]

What is the detail for the call. Calling number, called number, time of call etc

[Louis-Philippe Descamps]

oops sorry

calling: 000000

called: +3222249810

Time: 12:56 UK

Date: 10May

thanks

[Ayodeji Okanlawon]

It appears like the log you provided you have used + as the calling party mask..

Here is what I see in the logs for 12:56..

97527335.008 |12:56:44.425 |AppInfo  |Digit analysis: analysis results
97527335.009 |12:56:44.425 |AppInfo  ||PretransformCallingPartyNumber=+
|CallingPartyNumber=+
|DialingPartition=EU_Internal
|DialingPattern=+3222249810

---

97527376.003 |12:56:44.427 |AppInfo  |StationCdpc:    CcSetupReq - unicodeCallingPartyName='' asciiCallingPartyName='Unavailable' callingParty='+'

I expect in this case that a + is displayed on the phone..

I also see that you are doing restriction on the called number. I can see privacy on the called number..

SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.20.4.156:5060;branch=z9hG4bK06Bf1831c41dcd1ea14
From: "Unavailable" <sip:000000@va.sip.colt.net>;tag=gK06208635
To: <sip:+3222249810@10.20.44.21>;tag=76405267~691d352a-3cdb-4674-beac-9a44bf063354-147891208
Date: Tue, 10 May 2016 11:56:44 GMT
Call-ID: 1275506742_12098250@10.20.4.156
CSeq: 11651 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM10.5
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: "Brussel Test" <sip:+3222249810@10.20.44.21>
Privacy: id
Remote-Party-ID: "Brussel Test" <sip:+3222249810@10.20.44.21>;party=called;screen=yes;privacy=full
Contact: <sip:+3222249810@10.20.44.21:5060;transport=tcp>
Content-Length: 0

I need a log for when you only did caller id rectriction without any mask..

Please note that the logs should be taken only from the ff server..

gbsloccmlp0002.qbe.eo

[Louis-Philippe Descamps]

hi

removed the +

tested

calling: 000000

called: +3222249810

Time: 12:06 UK

Date: 12May

attached