Modify SIP Profiles

[danielsanmartin]

Hi

We have a CUBE and we would need to modify the following lines of the INVITE message sent to the SBC of the telephone provider:

Remote-Party-ID: "Daniel Sanmartin" <sip:+541198765432@X.X.X.X>;party=calling;screen=yes;privacy=off
From: "Daniel Sanmartin" <sip:+541198765432@X.X.X.X>;tag=DF143-6E3
Contact: <sip:+541198765432@X.X.X.X>

The correct format of those lines should be:

Remote-Party-ID: "Daniel Sanmartin" <sip:01198765432@X.X.X.X>;party=calling;screen=yes;privacy=off
From: "Daniel Sanmartin" <sip:01198765432@X.X.X.X>;tag=DF143-6E3
Contact: <sip:01198765432@X.X.X.X>

What should be the correct sip-profiles syntax to make these modifications?
What should be the correct command to apply sip-profiles to the dial-peer?

Can someone help?

Thanks

[b.winter]

IMHO, it doesn't make any sense, to ask for a fixed solution from anyone else. You won't learn anything by that. There are very good guides about SIP-profiles:
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-media-proxy.html

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-sip-param-mod.html

And also a Cisco Tool to test your rules before you configure them in the router:
https://cway.cisco.com/csa-new/#/sipprofiletest

[danielsanmartin]

Hi b.winter

Hope you are doing well!

I have already read those links and I can't find what I need to apply the requested rules.
I'm already using SIP Profile Rules within Collaboration Solutions Analyzer and I can't get it to work either.
Our telephone provider does not help us by indicating why the outgoing calls are not answered and I have to get Webex Callig working as soon as possible because the company cannot handle the outgoing calls that do not work.
Thank you very much for the help. I'll look elsewhere.

Regards

[Maren Mahoney]

Your SIP Profile would look something like this:

voice class sip-profiles 1
  request ANY sip-header From modify "sip:+5411(.*)@" "sip:011\1@"
  request ANY sip-header Remote-Party-ID modify "sip:+5411(.*)@" "sip:011\1@"
  request ANY sip-header Contact modify "sip:+5411(.*)@" "sip:011\1@"

As for where to implement the profile, that will depend in part on the dial-peer configuration in your CUBE. Certainly, if you apply it on the outbound dial-peer facing the service provider you should be fine. Be aware that this is changing the outbound messaging, so you will need to accommodate the inbound messaging as well.

I am unable to test, so please be sure to test before you implement.

Maren

[danielsanmartin]

Thank you Maren
The suggested modifications have not worked. The headers continue to arrive to the provider's SBC the same.
The provider has agreed to perform a debug on their part to verify why our calls are not arriving.
We are going to wait for the result of that debug to take the necessary actions

Regards.

[Roger Kallberg]

If you share your running configuration and the output from these debugs running simultaneously we should be able to see what’s going on in your SBC.

• debug ccsip message 
• debug voip ccapi inout 
• debug voip translation 

Make a call and get the output from the debugs and put it into a plain text file and the configuration in another file that you attach to your response.

In general if all you want to do is to change the called and calling numbers you should be able to do that with voice translation rules and profiles instead of using a SIP profile. Depending on the need to change various fields in the SIP dialogue you might need to use a SIP profile, but for the most it’s a little bit of overkill.

[danielsanmartin]

Hi Roger

I'm going to tell you what's happening. I'm setting up Webex Calling. At this time we cannot make outgoing calls. Incoming calls work correctly.
The telephone provider is not being of great help to us in solving the problem.
All the Cisco people consulted say that I should discuss the problem with the telephone provider.
In short, the debug of outgoing calls shows that 4 INVITEs are made to the IP address of the SBC of the telephone provider and then the call is canceled due to timeout. There is no response from the supplier or rejection message for the sent packages.
So, on my own, I'm trying to detect what the problem could be.
The supplier, in its few requirements, indicates the following:
2.3 OUTGOING CALLS
For outgoing calls, keep the following requirements in mind:
• The identification number (ANI) of each outgoing call must belong to the group of those provided by IPLAN; Any other ANI number other than those provided will be replaced by one considered as a header; except in the case of a special configuration requested in a particular case.
• All outgoing calls must be directed to the IP address 190.210.6.162 corresponding to the IPLAN SBC (Session Border Controller). For example, 0115021001@190.210.6.162
• SIP packets must include the Public IP Address of origin, since if they include the Private IP address exclusively they may be rejected.

We already have voice translation rules to meet these requirements, but the Remote-Party-ID, From and Contact headers still show the E164 format and not the one required by the provider (see first post with examples).
So, my idea was to change those headers to meet the provider's requirements and see if that way I can get outgoing calls to work correctly.

[Roger Kallberg]

From what I can see you’re translating the DNIS (called number), not ANI (calling number) as the service provider is specifying. Try checking the translation rules and make this change to the profile to have it apply the modifications to the calling number.

voice translation-profile SalientesIPLAN
 translate calling 10

Also you don’t need the destination and session target lines on dial peer 500 as it’s used in inbound direction. These commands are used on outbound dial peers. To remove them use these commands.

dial-peer voice 500 voip
 no destination-pattern
 no session target sip-server

[danielsanmartin]

Hi Roger.

dial-peer voice 500 voip and everything that starts with the number 5 belong to the alternative trunk and circuit for incoming and outgoing calls through the FXO ports. As they always say, what is working well should not be touched.

I would not be understanding what I should configure to modify ANI instead of NDIS.
translation-profile SalientesIPLAN is being applied in dial-peer voice 200 voip which is the one connected to the ITSP.

[Roger Kallberg · Accepted solution]

That’s up to you, IMHO having configuration that is not required is a bad habit.

On the topic of translation of the ANI, either use the commands I suggested in my previous response, ie this.

voice translation-profile SalientesIPLAN
 translate calling 10

Or create a new rule set to only change what you listed in your original post, then use this.

voice translation-rule 1
 rule 1 /^\+54(.*)/ /0\1/
!
voice translation-profile SalientesIPLAN
 translate calling 1

[Roger Kallberg]

For more information on how to configure translation rules and profiles please see this document. Configure Number Translation with Voice Translation Profiles 

[danielsanmartin]

There is already a voice translation-rule 10 command that formats all outgoing WxC numbers and adapts them to the format recognized by the ITSP. This is seen in the format of the To header in the INVITE that is made to the ITSP.
But, apparently, it only modifies the DNIS and does not modify the ANI

voice translation-rule 10
rule 1 /\+5411/ /011/
rule 2 /\+541115/ /01115/
rule 3 /\+548/ /08/
rule 4 /\+546/ /06/
rule 5 /\+542/ /02/
rule 6 /\+543/ /03/
rule 7 /\^10/ /01110/
rule 8 /\^11/ /01111/
rule 9 /\^12/ /01112/
rule 10 /\^13/ /01113/
rule 11 /\^14/ /01114/
rule 12 /911/ /011911/
rule 13 /999/ /011999/

[danielsanmartin]

Sorry Roger.
I already understand what you are trying to explain to me.
You are suggesting me to use translate calling. I am using translate called. Now I got it.

[danielsanmartin]

Ready.
The suggestion was made, but outgoing calls still do not work.
I think the only thing left is to keep praying so that the ITSP deigns to give us a little help, even if it's just that.