Display number in outgoing call

[servicios.internos]

Hi!

I need help to find out if it's possible to configure a displayed number eg 5555-551766 (this is a DID configure with num-exp) for outgoing call.

When users make external calls the displayed number is 5555-551700 and when customer wants call back our agents they can use the 5555-551766 number.

I'm trying to fix in 'Calling Party Transformations' in route pattern configuration but it wasn't works.

I put this in the 'Route Patterns' wasn't works.

Next I put this in the 'Directory Number' but still wasn't works.

The version of CUCM is 11.0.1.22900-14

I hope you can help me!

[Jaime Valencia]

Make sure you're hitting that RP and debug your GW to find out if you're sending the ANI you want or not.

[servicios.internos]

Debugged my GW, this is the file with the result. I'm not sure about nothing of this.

[R0g22]

Check dial-peer 10 on your GW. Does it have a translation ?

Incoming Leg, the ANI is correct. Once the outgoing leg is created, the ANI is being changed. Attach your router config is possible.

[servicios.internos]

This is my configuration.

The dial-peer 10 doesn't have a translation

[R0g22]

Ok. Your dial-peer 31 is configured with "clid network-number 5555551700" command which is overriding the ANI. Remove that command and then test. It should correct the ANI.

[servicios.internos]

It wasn't works. I don't know if I missed something

[R0g22]

Did you remove the command ?

[servicios.internos]

Yes, I did it

[R0g22]

Remove the network-provided command as well. I guess I missed that in my previous post.

[servicios.internos]

This is my configuration:

R1#show run | section dial-peer voice 31 voip
dial-peer voice 31 voip
description **OUTSIDE_TO_CELULARES**
translation-profile outgoing salida
destination-pattern 904[4-5]..........
session protocol sipv2
session target ipv4:x.x.x.x
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
dtmf-relay rtp-nte
codec g711ulaw

... didn't function

[R0g22]

Take a log with the following enabled -

debug ccsip message
debug voice ccapi inout
debug voip translation

[servicios.internos]

Here are the information.

By the way, thank you for help me.

[R0g22 · Accepted solution]

Post removing the commands, I don't see any issue now. Here is the SIP INVITE being sent to the ITSP. "From"/"Contact" headers have the correct info -

May 23 00:36:36.043: //1914897/0D944B000001/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:0445555555513@X.X.X.27:5060 SIP/2.0Via: SIP/2.0/UDP X.X.X.226:5060;branch=z9hG4bK4223F941From: "Bad guy" <sip:5555551766@X.X.X.226>;tag=4FF17C20-15D3To: <sip:0445555555513@X.X.X.27>Date: Wed, 23 May 2018 00:36:36 GMTCall-ID: 2EEAA79F-5D5811E8-BE97A33D-347B5593@X.X.X.226Supported: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0227822336-0000065536-0000074230-0169087404User-Agent: Cisco-SIPGateway/IOS-15.5.3.M3Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1527035796Contact: <sip:5555551766@X.X.X.226:5060>

Talk to your ITSP. Are they looking for a RPID/PAI as well ?

[servicios.internos]

My service provider fixed it. Thank you for all your help!!