CUBE - Copy SIP From header from Incoming INVITE to outgoing INVITE

[nuinoahmed]

Hi,

I have a CUBE between ITSP and CUCM.   I would like to modify the From header received in the incoming INVITE and use it in the outgoing INVITE

Received:
INVITE sip:+496xxxxxxxx46@192.168.1.10;user=phone SIP/2.0
Max-Forwards: 68
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: "+496xxxxxxxx46" <sip:+496xxxxxxxx46@192.168.1.10;user=phone>
From: "+316xxxxxxx7" <sip:+316xxxxxxx7@212.10.1.34>;tag=3821351883-1068052432
P-Asserted-Identity: "+4930xxxxxxx10" <sip:+4930xxxxxxx10@212.10.1.34;user=phone>
Call-ID: 264551051-3821351883-200638263@SBC1-ABC-EU.exemple.com
CSeq: 1 INVITE
Allow: MESSAGE,PRACK,INFO,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 212.10.1.34:5060;branch=z9hG4bKb005fb6fc6c3f229959ef2d648333f2c
Contact: <sip:+316xxxxxxx7@212.10.1.34:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 347
v=0
o=SBC2-ABC-EU 2496788555450729 1 IN IP4 212.10.1.34
s=sip call
c=IN IP4 212.10.1.35
t=0 0
m=audio 65102 RTP/AVP 8 0 18 96 13 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 G726-32/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Sent:
INVITE sip:+496xxxxxxxx46@10.218.11.13 SIP/2.0
Via: SIP/2.0/TCP 10.10.128.66:5060;branch=z9hG4bK1A325E32C
From: "+4930xxxxxxx10" <sip:+4930xxxxxxx10@10.10.128.66>;tag=96988838-F0A
To: <sip:+496xxxxxxxx46@10.218.11.13>
Date: Wed, 03 Feb 2021 13:41:57 GMT
Call-ID: 6B783808-655C11EB-9F6CFE9F-76F149F1@10.10.128.66
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 600
Cisco-Guid: 1803010717-1700532715-2674327199-1995524593
User-Agent: Cisco-SIPGateway/IOS-16.6.4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1612359717
Contact: <sip:+4930xxxxxxx10@10.10.128.66:5060;transport=tcp>
Call-Info: <sip:10.10.128.66:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 67
P-Asserted-Identity: "+4930xxxxxxx10" <sip:+4930xxxxxxx10@10.10.128.66>
Session-ID: d6e6a4ea31b55239a00a08ee52f06598;remote=00000000000000000000000000000000
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 298
v=0
o=CiscoSystemsSIP-GW-UserAgent 7598 4889 IN IP4 10.10.128.66
s=SIP Call
c=IN IP4 10.10.128.66
t=0 0
m=audio 45722 RTP/AVP 8 0 101 19
c=IN IP4 10.10.128.66
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

I tried following but it failed

voice class sip-copylist 50
sip-header FROM

!

dial­peer voice 100 voip
voice class sip-copylist 50

!

voice class sip-profiles 50
request INVITE peer-header sip From copy "sip:(.*)@" u01
request INVITE sip-header From modify ".*@(.*)" "INVITE sip:\u01@\1"

!

dial-peer voice 100 voip        <<<<<<  FROM ITSP
voice class sip-copylist 50

!

dial-peer voice 400 voip        <<<<<<<  To CUCM
voice-class sip profiles 50

Can someone review the above and advise what went wrong and if I missed something? 

Thanks in advance

[Roger Kallberg · Accepted solution]

Have a look at this post, it sounds like it should be similar to what you ask.

https://community.cisco.com/t5/ip-telephony-and-phones/cube-sip-profile-amp-copy-list-feature-error-when-handling-from/td-p/4270906

[nuinoahmed]

Thanks Roger,

I prepared the config I think correct now.  I will apply it tomorrow to the CUBE and test.  Will let you know.

[Roger Kallberg]

If you want to pre test the SIP profile I recommend you to look through this document about the test tool for SIP profiles that Cisco have provided for us. https://community.cisco.com/t5/collaboration-voice-and-video/sip-profile-test-tool/ta-p/3162632

[nuinoahmed · Accepted solution]

Hi Roger,

Finally I got it working. Thanks for your help.

It was not an easy one as the SIP Profiles Test Tool may show you as working while you still have an extra \ at the end of the number.  So on cube this will of course not work.  

Here is the working config, it may help and save some time for others:

voice class sip-profiles 10
request INVITE peer-header sip From copy "<sip:(.*)@" u01
request INVITE sip-header From modify ".*@(.*)" "From: \"\u01\" <sip:\u01@\1"
request INVITE sip-header P-Asserted-Identity modify ".*@(.*)" "P-Asserted-Identity: \"\u01\" <sip:\u01@\1"
!
voice class sip-copylist 10
sip-header From

!

dial-peer voice 50 voip    <<<<<<<<<<<<<<  From ITSP

voice-class sip copy-list 10

!

dial-peer voice 100 voip    <<<<<<<<<<<<<<   To CUCM

voice-class sip profiles 10

[Roger Kallberg]

Glad you managed to get it work. One question, was it intentional that you marked your own reply as the answer to your post?

[nuinoahmed]

No, that was not intentional.  your first reply did put me really in the right direction.