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Beginner

Calling ID missing, and show "Private" on screen

Hi everybody, 

CUCM is using QSIG to connect with Siemens via VG

Call flow : Cisco IP phone --- CUCM --- VG --- Siemens PABX --- Siemens Phone

When "Cisco User A" 171 calls to "Siemens Phone X" (700), it does not show the caller ID "Siemens Phone X" on Cisco IP phone.

It only shows "Private" on cisco phone screen

When looking at the sip signal 

The name "Siemens Phone X" does appear in incoming signal "SIP/2.0 183 Session Progress

However, "Siemens Phone X" disappear in incoming signal "SIP/2.0 200 OK", and all the rest of the SIP signal

Here are the SIP signal between VG and Siemens PABX

15346755.002 |15:34:29.910 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.141.101.201 on port 5060 index 790
[1569568,NET]
INVITE sip:700@10.141.101.201:5060 SIP/2.0
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf0996fdd4759
From: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
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:10.141.101.203:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3903682048-0000065536-0000000658-3412430090
Session-Expires: 1800
P-Asserted-Identity: "Cisco User A" <sip:171@10.141.101.203>
Remote-Party-ID: "Cisco User A" <sip:171@10.141.101.203>;party=calling;screen=yes;privacy=off
Contact: <sip:171@10.141.101.203:5060;transport=tcp>;video;audio
Max-Forwards: 69
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 67

08020292 05a10403 8090a318 03a98381
1c2291aa 06800100 820100a1 17020100
06042b0c 0900800c 47696c62 65727420
576f6e67 6c050081 31373170 04803730
300d0a


-------------------------
15346763.002 |15:34:29.918 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.141.101.201 on port 5060 index 790 with 423 bytes:
[1569570,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf0996fdd4759
From: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0

--------

15346776.007 |15:34:30.003 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.141.101.201 on port 5060 index 790 with 1526 bytes:
[1569573,NET]
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf0996fdd4759
From: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>;tag=91B0C5CC-145
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Siemens User X" <sip:700@10.141.101.201>;party=called;screen=no;privacy=off
Contact: <sip:700@10.141.101.201:5060;transport=tcp>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 735

--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=CiscoSystemsSIP-GW-UserAgent 2670 6495 IN IP4 10.141.101.201
s=SIP Call
c=IN IP4 10.141.101.201
t=0 0
m=audio 30398 RTP/AVP 0 8 18 101
c=IN IP4 10.141.101.201
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--uniqueBoundary
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 53

08020ebf 011c2491 aa068001 00820100
8b0100a1 16020219 bb06042b 0c090182
0a426172 72696520 4875691e 02808895
3201810d 0a
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
-------------------------------------


15346811.007 |15:34:32.877 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.141.101.201 on port 5060 index 790 with 1595 bytes:
[1569575,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf0996fdd4759
From: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>;tag=91B0C5CC-145
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:700@10.141.101.201>;party=called;screen=no;privacy=off
Contact: <sip:700@10.141.101.201:5060;transport=tcp>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 738

--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=CiscoSystemsSIP-GW-UserAgent 2670 6495 IN IP4 10.141.101.201
s=SIP Call
c=IN IP4 10.141.101.201
t=0 0
m=audio 30398 RTP/AVP 0 8 18 101
c=IN IP4 10.141.101.201
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--uniqueBoundary
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 56

08020ebf 071c2491 aa068001 00820100
8b0100a1 16020219 bd06042b 0c090382
0a426172 72696520 4875694c 0509a337
30309532 01810d0a
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional

---------------------------
15346871.001 |15:34:32.885 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.141.101.201 on port 5060 index 790
[1569576,NET]
ACK sip:700@10.141.101.201:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf09a657811bd
From: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>;tag=91B0C5CC-145
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 239

v=0
o=CiscoSystemsCCM-SIP 482740 1 IN IP4 10.141.101.203
s=SIP Call
c=IN IP4 10.141.1.239
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26372 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

-------------------------

15346914.009 |15:34:32.891 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.141.101.201 on port 44111 index 791 with 1003 bytes:
[1569579,NET]
INFO sip:171@10.141.101.203:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.141.101.201:5060;branch=z9hG4bK183FF7D8
From: <sip:700@10.141.101.201>;tag=91B0C5CC-145
To: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
Date: Tue, 12 Jan 2016 07:34:32 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Max-Forwards: 70
Timestamp: 1452584072
CSeq: 101 INFO
Contact: <sip:700@10.141.101.201:5060;transport=tcp>
Remote-Party-ID: <sip:700@10.141.101.201>;party=called;screen=no;privacy=off
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 311

--uniqueBoundary
Content-Type: application/qsig
Content-Disposition: signal;handling=optional
Content-Length: 43

00000000 00000000 00000000 00000000
00000000 00000000 00000000 00000000
00000000 00000000 000000

---------------------------

15346921.001 |15:34:32.892 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.141.101.201 on port 44111 index 791
[1569580,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.141.101.201:5060;branch=z9hG4bK183FF7D8
From: <sip:700@10.141.101.201>;tag=91B0C5CC-145
To: "Cisco User A" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
Date: Tue, 12 Jan 2016 07:34:32 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
CSeq: 101 INFO
Contact: <sip:171@10.141.101.203:5060;transport=tcp>
Content-Length: 0

Is there anything I can enable the Calling ID display on Cisco IP phone?

Thanks in advance


Sam

Everyone's tags (3)
1 ACCEPTED SOLUTION

Accepted Solutions
VIP Advisor

This has to be done on the

This has to be done on the Siemens side. Nothing to be done on Cisco side. Worth posing the question on Siemens forum

9 REPLIES 9
VIP Advisor

This has to be done on the

This has to be done on the Siemens side. Nothing to be done on Cisco side. Worth posing the question on Siemens forum

Beginner

Hi ,

Hi ,

Thanks for the reply

I believe the VG has passed the "Name" field to CUCM, otherwise CUCM wont send the "Name" to IP phone in SIP

Attached 2 SIP signal diagram

Pic 1 : during the "Ringing" (from CUCM to IP Phone)
Remote-Party-ID: "Barrie xxxxxx" <sip:700@10.141.101.203>;party=called;screen=no;privacy=name

Pic 2 : during " OK " (from CUCM to IP phone)
Remote-Party-ID: "Barrie xxxxxx" <sip:700@10.141.101.203>;party=called;screen=yes;privacy=full


According to other forum topic (https://supportforums.cisco.com/discussion/12621196/cucm-86-sip-trunk-cube-sip-trunk-ringing-connected-party-display), privacy=full (this means "Private")

Is there any way to config/aware in CUCM which can change privacy=no?

Thanks in advance


Sam

VIP Mentor

We dont have the full picture

We dont have the full picture here. What does the gateway send to CUCM for the 200 OK. We need to know why CUCM is changing the remote-party-id privacy to full. I dont think CUCM will just change RPID just like that. Please post the output of the gateway logs or have a look yourself. If the gateway is sending privacy=full, then check if this is what the Siemens is sending to the gateway..

Please rate all useful posts
Beginner

Thanks for the reply, 

Thanks for the reply, 

If want to check the gateway output, the debug should be 

debug ccsip message 

and any other debug are needed?

Thanks in advance

Highlighted
Beginner

Finally, CiscoTAC responded

Finally, CiscoTAC responded the 3rd Party PBX has restricted the calling ID signal to CUCM. 

That's why CUCM ip phone cannot read the ID

Beginner

For the signal #3 and #4, it

For the signal #3 and #4, it makes me confusing

1. VG > CUCM-sub

Remote-Party-ID has VALUE

2. CUCM-Sub > IP phone

Remote-Party-ID has VALUE

3. VG > CUCM-sub

Remote-Party-ID's VALUE is disappeared?

4. CUCM-Sub > IP phone

Remote-Party-ID's VALUE has value, but privacy=full

VIP Mentor

You are right. The gateway

You are right. The gateway sends caller Id.. Can you send us cucm traces? We need to know why cucm is setting privacy to full 

Please rate all useful posts
Beginner

Here is the CUCM trace,

Here is the CUCM trace, thanks for the help

VIP Mentor

Sam,

Sam,

after looking at the logs here is what is going on...

+++ here we see CUCM sending 180 ringing to the IP Phone with the called name +++

15346802.001 |15:34:30.006 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.141.1.239 on port 49942 index 470
[1569574,NET]
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.141.1.239:49942;branch=z9hG4bK196b5ad4
From: "Gilbert Wong" <sip:171@10.141.101.203>;tag=c4729551a6f33ad661a4e6b8-14012ed9
To: <sip:7@10.141.101.203>;tag=482736~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339949
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: c4729551-a6f30d4d-348f8034-50f07dcb@10.141.1.239
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= 2-102154; call-instance= 1
Send-Info: conference, x-cisco-conference
Remote-Party-ID: "Barrie Hui" <sip:700@10.141.101.203>;party=called;screen=no;privacy=name
Remote-Party-ID: <sip:700@10.141.101.203;user=phone>;party=x-cisco-original-called;privacy=off
Contact: <sip:7@10.141.101.203:5060;transport=tcp>
Content-Length: 0

+++ Now when we get the 200 OK from the gateway, observe that the called name is no longer present , only the called number +++
---
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.141.101.203:5060;branch=z9hG4bKf0996fdd4759
From: "Gilbert Wong" <sip:171@10.141.101.203>;tag=482740~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339950
To: <sip:700@10.141.101.201>;tag=91B0C5CC-145
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: e8ad7600-6941ac85-3b17-cb658d0a@10.141.101.203
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:700@10.141.101.201>;party=called;screen=no;privacy=off
Contact: <sip:700@10.141.101.201:5060;transport=tcp>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 738

---
+++ Next CUCM updated the called identity name, observer that after the update we no longer have the remote calling name (called name)
15346814.004 |15:34:32.878 |AppInfo  |SIPCdpc(10513) - processRemoteIdentityInfo: name=,remoteCnName=Barrie Hui,After update remoteCnName=;

Next CUCM sends a 200 OK and withheld the called name using the privacy=full and only sent the called number...


SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.141.1.239:49942;branch=z9hG4bK196b5ad4
From: "Gilbert Wong" <sip:171@10.141.101.203>;tag=c4729551a6f33ad661a4e6b8-14012ed9
To: <sip:7@10.141.101.203>;tag=482736~bdbfb3e7-cc54-4caf-a277-532bc75a742e-41339949
Date: Tue, 12 Jan 2016 07:34:29 GMT
Call-ID: c4729551-a6f30d4d-348f8034-50f07dcb@10.141.1.239
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; gci= 2-102154; call-instance= 1
Send-Info: conference, x-cisco-conference
Remote-Party-ID: "Barrie Hui" <sip:700@10.141.101.203>;party=called;screen=yes;privacy=full
Remote-Party-ID: <sip:700@10.141.101.203;user=phone>;party=x-cisco-original-called;privacy=off
Contact: <sip:7@10.141.101.203:5060;transport=tcp>
Content-Type: application/sdp

So it does look like the issue is from your PBX..Why is it stripping the called name from the 200 OK..

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