How to do SIP normalization, CME?

Unanswered Question
May 4th, 2012

Hi all,

I'm facing the problem, that our SIP provider is sending the wrong number (registration nr.) in the SIP FROM header.

Check my other discussion:

https://supportforums.cisco.com/message/3586648#3586648

Now I'm trying to copy the SIP To: information to From: with sip profile, but somehow it can't copy it.

Here my setup:

Cisco 2821, c2800nm-adventerprisek9_ivs-mz.151-4.M4.bin, CME Version 8.6

voice call send-alert

voice call convert-discpi-to-prog

voice disc-pi-incoming-on

!

voice service voip

no ip address trusted authenticate

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

h323

  h245 caps mode restricted

sip

  asserted-id ppi

!

!

voice class uri TRUNK sip

user-id 0815440096

....

voice class sip-profiles 102

request INVITE sip-header To copy "<sip:(.*)@(.*)>" u01

request INVITE sip-header From modify "<sip:.*@(.*)>" "<sip:\u01@\1>"

....

voice class sip-copylist 1

sip-header To

...

voice translation-rule 1

rule 1 // // plan any unknown

!        

voice translation-rule 2

rule 1 // // plan any unknown

!

voice translation-rule 40

rule 2 /\(.*\)/ /0\1/

!        

voice translation-rule 190

rule 1 /^0\(.*\)/ /\1/

rule 2 /^9\(.*\)/ /\1/

!        

voice translation-rule 191

rule 1 /297/ /0815440097/

rule 2 /296/ /0815440096/

rule 3 /^0\(.*\)/ /\1/

!

voice translation-rule 192

rule 2 /^0815440097/ /297/

rule 3 /^0815440096/ /296/

voice translation-profile TP_IN_SIP

translate calling 40

translate called 192

!

voice translation-profile TP_IN_SIP_1

translate calling 40

translate called 1

!

voice translation-profile TP_OUT_SIP

translate calling 191

translate called 190

! -> INCOMING PEER FROM SIP-PROVIDER to CME

dial-peer voice 2000 voip

description ** SIP TRUNK IN **

b2bua

session protocol sipv2

session target dns:sip12.e-fon.ch

session transport udp

incoming called-number .

incoming uri to TRUNK

voice-class sip profiles 102

voice-class sip copy-list 1

dtmf-relay rtp-nte

codec g711alaw

no vad

!

! -> INCOMING PEER FROM DIAL-PEER 2000 to PHONE

dial-peer voice 2010 voip

description ** MATCH-URI->changes doing here **

translation-profile incoming TP_OUT_SIP

translation-profile outgoing TP_IN_SIP_1

shutdown

destination-pattern 081544009.

b2bua

session protocol sipv2

session target ipv4:192.168.115.1

session transport udp

incoming called-number .

voice-class sip profiles 102

dtmf-relay rtp-nte

codec g711alaw

no vad  

!

!        

sip-ua

credentials username 0815440096 password 7 xxxx realm sip21.e-fon.ch

keepalive target dns:sip12.e-fon.ch

authentication username 0815440096 password 7 xxxx

no remote-party-id

retry invite 2

retry response 2

retry bye 2

retry register 2

retry options 1

registrar dns:sip12.e-fon.ch expires 60

sip-server dns:sip12.e-fon.ch

Any ideas? Does the copy function works on the CME or do I need a CUBE?

http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/

Thanks,

Norbert

I have this problem too.
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alig.norbert Fri, 05/04/2012 - 00:25

You are right, but that provider changed the SIP FROM information as a "security" feature and are not willing to change it back

Sent from Cisco Technical Support iPhone App

Ayodeji oladipo... Fri, 05/04/2012 - 01:05

Norbert,

I have l ooked at your other trace and it looks a bit confusing to me..

Can you do another test call...and send only the following debug along with the calling and called number..

Please confirm that you sip provder is 212.55.198.134 and your cme is 192.168.70.240

debug voip ccapi inout

debug ccsip messages

Please remove the other debugs...

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Posted May 4, 2012 at 12:10 AM
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