P-Associated URI as P-Preferred-Identity on CUBE

[mcaldogne]

Hello!

We are deploying a new CUBE with multiple registrations to the ITSP, but have a problem.

The ITSP ask if it is possible to send the "P-Associated URI" in the "P-Asserted-Identity" field for outgoing calls.

Actually we have the same value as in the contact field, 

Do you have an idea? 

Thanks, Mirko

[b.winter]

And what is your question now?
How to enable the PAI header?

[mcaldogne]

The PAI Header is still enable but it is not working correctly for the ITSP.

The question if it possible to put the value P-Associated URI that is in the registration, also in the field P-Preferred-Identity of the SIP message.

[b.winter]

You can use a sip-profile to modify the value of the PAI header to whatever you want.
Create a sip-profile, add a rule for the PAI modification and assign it to the outbound dial-peer towards ITSP.

But why doesn't he like to have a different number in there? Assuming you have the main number 044422334 and a DID range of 1 digit, normally all numbers from 0444223340 to 0444223349 are allowed in the PAI header. I have never seen a ITSP not allowing this, because all those numbers belong to you and the SIP trunk.

[mcaldogne]

OK thanks you, I tested with a new custom SIP profile!

I don't know exactly why ITSP want that, but I think the reason is related to the multiple registrations from one CUBE.

[b.winter]

SIP providers^^
And no, it shouldn't have anything to do with multiple SIP trunks on your CUBE.

[Roger Kallberg · Accepted solution]

If all you'd like to do is replace the value in the PAI with a fixed value you can easily do that with a SIP profile. Something along with this, that you put as outbound on the dial peer towards your ITSP should do.

voice class sip-profiles 10
 request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422334\2" 
 response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422334\2"

[mcaldogne]

Thanks for the information, it works with your settings in the SIP profile. 

But now I have another problem...

I have multiple tenant and I thought it would be possible to have for each tenant dedicated dial-peers (each dial-peer with the own SIP profile). The problem is that all the outgoing calls are matching with the first available dial-peer, without taking into consideration the tenant. For example I would expect that tenant 110 should match dial-peer 11  and tenant 120 sould match dial-peer 12 for calls to +.  Am I doing something wrong?

voice class sip-profiles 11
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422334\2" 
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422334\2" 

voice class sip-profiles 12
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422335\2" 
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*<sip:).*(@.*>)" "P-Asserted-Identity: \1044422335\2"

voice class tenant 110
registrar dns:sip.xxxxx.xx:5060 expires 3600
credentials username 044422334 password <password> realm sip.xxxxx.xx
authentication username 044422334 password <password>
sip-server dns:sip.xxxxx.xx:5060
no pass-thru content custom-sdp
!
voice class tenant 120
registrar dns:sip.xxxxx.xx:5060 expires 3600
credentials username 044422335 password <password> realm sip.xxxxx.xx
authentication username 044422335 password <password>
sip-server dns:sip.xxxxx.xx:5060
no pass-thru content custom-sdp

dial-peer voice 11 voip
translation-profile outgoing OUTGOING
destination-pattern +T
session protocol sipv2
session target dns:sip.xxxxx.xx
voice-class codec 1 
voice-class sip profiles 11
voice-class sip tenant 110
no voice-class sip session refresh
dtmf-relay rtp-nte
no fax-relay sg3-to-g3
fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

dial-peer voice 12 voip
translation-profile outgoing OUTGOING
destination-pattern +T
session protocol sipv2
session target dns:sip.xxxxx.xx
voice-class codec 1 
voice-class sip profiles 12
voice-class sip tenant 120
no voice-class sip session refresh
dtmf-relay rtp-nte
no fax-relay sg3-to-g3
fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

[b.winter]

The tenant has nothing to do with dial-peer matching.
You could use dial-peer groups assigned to the incoming dial-peer, to control which outgoing dial-peers should be considered --> Check the docs how to configure.

Or you get you dial-peer matching right.

[mcaldogne]

I understand, thank you for your reply!

I will check the docs beacuse I have no experience with dial-peers groups.

[Roger Kallberg]

This document should provide you with the needed information. In Depth Explanation of Cisco IOS and IOS-XE Call Routing 

[mcaldogne]

Thank you for your help!