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Restricting Caller ID on SIP trunk to PRI gateway

Sean McQuade
Level 5
Level 5

Hi all,

 

I am having a problem restricting outgoing CLID on calls made via a SIP trunk from CUCM to a PRI gateway.

 

We were previously on CUCM v7 and have now upgraded to 10.5 and as part of this, have converted all our H323 gateways to SIP. We have one branch office which wants to restrict the CLID on all outgoing calls, and previously this was just set to restrict on the H323 gateway config within Call Manager and seemed to work correctly - with the SIP trunk, I cannot get this to work.

 

I have tried to restrict it on the trunk config, using a calling party transform mask, and also by adding "clid restrict" to the incoming VOIP dial-peer from CUCM and the outgoing POTS dial-peer to the PSTN.

 

The SIP leg of the call seems to come in with anonymous in the invite, but has the PAI and Remote party, so I'm wondering if this is potentially causing the issue. In this case I am calling a UK mobile (0044XXX) - the number ending in 80 is the calling number:

 

*Jan 19 15:02:38.274: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0044XXXXXX7143@10.19.11.253:5060 SIP/2.0
Via: SIP/2.0/TCP 10.19.99.241:5060;branch=z9hG4bK161553e57dee3
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2871972~cc22c68e-4888-4e0d-972e-1fcb1cce5a16-79960756
To: <sip:0044XXXXXX7143@10.19.11.253>
Date: Mon, 19 Jan 2015 15:06:19 GMT
Call-ID: b79cad00-4bd11d6b-14447-f163130a@10.19.99.241
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.19.99.241:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 3080498432-0000065536-0000000839-4049801994
Session-Expires:  1800
P-Asserted-Identity: <sip:+4122XXXXX80@10.19.99.241>
Privacy: id
Remote-Party-ID: <sip:+4122XXXXX80@10.19.99.241>;party=calling;screen=yes;privacy=full
Contact: <sip:10.19.99.241:5060;transport=tcp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 358

v=0
o=CiscoSystemsCCM-SIP 2871972 1 IN IP4 10.19.99.241
s=SIP Call
c=IN IP4 10.19.20.6
b=TIAS:64000
b=AS:64
t=0 0
m=audio 22084 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:20
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

*Jan 19 15:02:38.282: //-1/B79CAD000000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=+4122XXXXX80
   ----- ccCallInfo IE subfields -----
   cisco-ani=sip:anonymous@anonymous.invalid
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=1
   cisco-anisi=1
   dest=sip:0044XXXXXX7143@10.19.11.253:5060
   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

*Jan 19 15:02:38.286: //-1/B79CAD000000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x8A55A8E0, Call Info(
   Calling Number=sip:anonymous@anonymous.invalid,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted),
   Called Number=sip:0044XXXXXX7143@10.19.11.253:5060(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=4000, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=38491
*Jan 19 15:02:38.286: //-1/B79CAD000000/CCAPI/ccCheckClipClir:
   In: Calling Number=sip:anonymous@anonymous.invalid(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted)
*Jan 19 15:02:38.286: //-1/B79CAD000000/CCAPI/ccCheckClipClir:
   Out: Calling Number=sip:anonymous@anonymous.invalid(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted)
*Jan 19 15:02:38.286: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

 

This part appears to be okay, and the caller ID is not shown. It then matches the outgoing POTS dial-peer and still appears to be restricted:

 

*Jan 19 15:02:38.290: //38491/B79CAD000000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=2001, Params=0x8BE5E150, Progress Indication=NULL(0)
*Jan 19 15:02:38.290: //38491/B79CAD000000/CCAPI/ccCheckClipClir:
   In: Calling Number=sip:anonymous@anonymous.invalid(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted)
*Jan 19 15:02:38.290: //38491/B79CAD000000/CCAPI/ccCheckClipClir:
   Out: Calling Number=sip:anonymous@anonymous.invalid(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted)

 

But then just underneath this:

 

*Jan 19 15:02:38.290: //38491/B79CAD000000/CCAPI/ccCallSetupRequest:
   Calling Number=sip:anonymous@anonymous.invalid(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Restricted),
   Called Number=0044XXXXXX7143(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=+4122XXXXX80, Final Destination Flag=TRUE,
   Guid=B79CAD00-0001-0000-0000-0347F163130A, Outgoing Dial-peer=2001
*Jan 19 15:02:38.290: //38491/B79CAD000000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=+4122XXXXX80

 

I'm not sure where the "Account Number" is coming from and then sends the call to the PSTN via the BRI, with the calling number populated:

 

*Jan 19 15:02:38.294: :FEATURE_VSA attributes are: feature_name:0,feature_time:2272865472,feature_id:6711
*Jan 19 15:02:38.294: //38492/B79CAD000000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1
*Jan 19 15:02:38.294: //38492/B79CAD000000/CCAPI/ccCallSetContext:
   Context=0x8BE5E100
*Jan 19 15:02:38.294: //38491/B79CAD000000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2001
*Jan 19 15:02:38.294: ISDN BR2 Q931: Applying typeplan for sw-type 0x1 is 0x0 0x0, Calling num +4122XXXXX80
*Jan 19 15:02:38.298: ISDN BR2 Q931: Sending SETUP  callref = 0x004F callID = 0x8A88 switch = basic-net3 interface = User
*Jan 19 15:02:38.298: ISDN BR2 Q931: TX -> SETUP pd = 8  callref = 0x4F
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0x82
                Preferred, B2
        Calling Party Number i = 0x00A1, '+4122XXXXX80'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x80, '0044XXXXXX7143'
                Plan:Unknown, Type:Unknown

 

We have another site in Poland where the "clid restrict" on the outgoing POTS dial-peer appears to work correctly. Is there a possibility that this is a carrier issue, or is there anything else I can change?

 

Thanks

Sean

1 Reply 1

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

There are a few things you can try even though from the logs everything should work..

1. Restrict caller id once privacy exists

conf t

voice service voip

clid strip pi-restrict all

2. Disable remote-party-id globally (this way only the p-asserted identity will be used for cli)

conf t

sip-ua

no remote-party-id

Test again

and please attach logs

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