07-21-2015 08:13 AM - edited 03-17-2019 03:42 AM
Hi all
I have a problem to modify the P-Asserted-Identity with my Cisco CUBE.
I want to change the value from <sip:+41yyyyyyyyy@xxx.xxx.xxx.xxx> to <tel:+41yyyyyyyyy@xxx.xxx.xxx.xxx>
I have the following SIP Profile:
voice class sip-profiles 100
request ANY sip-header P-Asserted-Identity modify "<sip:(.*)@.*>" "<tel:\1>"
and this is my Dial-Peer:
dial-peer voice 2100 voip
description *** dial-peer Provider ***
translation-profile incoming FROM-PSTN
destination-pattern +T
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
session transport tcp
incoming called-number +41yyyyy....
voice-class codec 1
voice-class sip srtp-auth sha1-32
voice-class sip profiles 100
dtmf-relay rtp-nte
no vad
Can anybody explain me why the Cube doesn't modify my P-ID?
Thank you very much for your help.
Cheers,
Pascal
Jul 21 15:00:23.819: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+41aaaaaaaaa@zzz.zzz.zzz.zzz:5060 SIP/2.0
Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKf35d6658b3e7
From: "Pascal Blumenthal" <sip:+41yyyyyyyyy@xxx.xxx.xxx.xxx>;tag=182794~59407172-9872-5f91-8b80-5d18f687c0d0-66050726
To: <sip:+41795985590@10.250.1.96>
Date: Tue, 21 Jul 2015 15:00:23 GMT
Call-ID: 3503a300-5ae15e87-ef54-3301fa0a@10.250.1.51
Supported: 100rel,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
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 0889430784-0000065536-0000000210-0855767562
Session-Expires: 1800
P-Asserted-Identity: "Firstname Surname" <sip:+41yyyyyyyyy@xxx.xxx.xxx.xxx>
Remote-Party-ID: "Firstname Surname" <sip:+41yyyyyyyyy@xxx.xxx.xxx.xxx>;party=calling;screen=yes;privacy=off
Contact: <sip:+41yyyyyyyyy@xxx.xxx.xxx.xxx:5060;transport=tcp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 200
Solved! Go to Solution.
07-21-2015 11:59 PM
I just tested this in my lab and got it working. Make sure that your are matching the correct outgoing dialpeer. Please post the entire config and the output of debug voice dialpeer all.
Config
voice class sip-profiles 1
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:(.*)" "P-Asserted-Identity:\1<tel:\2"
!
dial-peer voice 717 voip
destination-pattern xxxxxxxxxx
session protocol sipv2
session target ipv4:x.x.x.x
voice-class sip asserted-id pai
voice-class sip profiles 1
Debug
Jul 22 2015 06:52:32.505 UTC: //62893/388D3D800004/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:xxxxxxxxxx@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK5743FC2
From: "Mohammed Al Baqari" <sip:xxxxxxx@x.x.x.x>;tag=FBF1FF60-21A8
To: <sip:xxxxxxx@x.x.x.x>
Date: Wed, 22 Jul 2015 06:52:32 GMT
Call-ID: FA7FF08-2F7511E5-B174EFBA-912C2449@10.170.170.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0948780416-0000065536-0000262786-0184855050
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1437547952
Contact: <sip:xxxxxxx@x.x.x.x:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: "Mohammed Al Baqari" <tel:xxxxxxx@x.x.x.x>
Session-Expires: 1800
Content-Length: 0
07-21-2015 09:11 AM
You configuration should look as follow:
voice class sip-profiles 100
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<)sip:(.*)" "P-Asserted-Identity:\1tel:\2"
07-21-2015 11:25 PM
Hi Mohammed
Thank you for your answer but unfortenately, my CUBE doesn't change any value in the P-Asserted Identity Field.
Any other suggestions?
Best regards,
Pascal
07-21-2015 11:59 PM
I just tested this in my lab and got it working. Make sure that your are matching the correct outgoing dialpeer. Please post the entire config and the output of debug voice dialpeer all.
Config
voice class sip-profiles 1
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:(.*)" "P-Asserted-Identity:\1<tel:\2"
!
dial-peer voice 717 voip
destination-pattern xxxxxxxxxx
session protocol sipv2
session target ipv4:x.x.x.x
voice-class sip asserted-id pai
voice-class sip profiles 1
Debug
Jul 22 2015 06:52:32.505 UTC: //62893/388D3D800004/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:xxxxxxxxxx@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK5743FC2
From: "Mohammed Al Baqari" <sip:xxxxxxx@x.x.x.x>;tag=FBF1FF60-21A8
To: <sip:xxxxxxx@x.x.x.x>
Date: Wed, 22 Jul 2015 06:52:32 GMT
Call-ID: FA7FF08-2F7511E5-B174EFBA-912C2449@10.170.170.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0948780416-0000065536-0000262786-0184855050
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1437547952
Contact: <sip:xxxxxxx@x.x.x.x:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
P-Asserted-Identity: "Mohammed Al Baqari" <tel:xxxxxxx@x.x.x.x>
Session-Expires: 1800
Content-Length: 0
07-22-2015 03:58 AM
Hi Mohammed
You are right...it works. I was looking at the wrong dial-peer.
Thank you very much for your help.
Regards,
Pascal
06-16-2016 06:46 AM
Hi Mohammed,
I just have the same problem that I want to modify the "P-Asserted-Identity" as the provider expect a special value.
The "normal" unmodified P-Asserted-Identity looks like this
P-Asserted-Identity: "username" <sip:XXXX77@x.x.x.x>
I want to change this P-Asserted-Identity to this one
P-Asserted-Identity: "username"<sip:YYYYYYYY77@y.y.y.y:5060>
X = internal DN
x = internal IP of CUCM
Y = external DN
y = external IP
I tried to do this with this config
voice class sip-profiles 1
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
dial-peer voice 11 voip
description SIP-Outgoing
translation-profile incoming SipInbound
translation-profile outgoing SipOutbound
destination-pattern 0T
modem passthrough nse codec g711alaw
session protocol sipv2
session target ipv4:y.y.y.y
session transport udp
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay sip-notify rtp-nte
codec g711alaw
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
In the log I also see that dialpeer 11 should be used:
*Jun 16 15:16:31.242 MESZ: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=11 Is Matched
But the problem is that the P-Asserted-Identity is not modified and still the default value.
Would did I wrong?
BR
Michael
07-09-2024 11:54 AM
hello all,
I am new to SIP profiles, so I am trying to only change the username in the P-asserted-identity field.
thanks
07-09-2024 01:23 PM
As this post is marked as solved and is quite old it would be advisable for you to create your own post to ask your question. When doing so please provide as clear and detailed information as you can on what you’re looking to achieve so that it’s easier for us to provide you help.
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