Cisco’s “Route Next Hop By Calling Party Number” translation pattern option has addressed the common question of “how do I route or block calls based on Caller ID?” since CUCM version 8. However, there still remains a limitation and that is how to deal with “Anonymous”, “Restricted” or “Unknown” Calling Numbers. The problem is that the SIP information actually lists this as part of the From header where CUCM typically would expect to see a numeric value. As an example, talk a look at this SIP invite:
INVITE sip:firstname.lastname@example.org:61482 SIP/2.0 Via: SIP/2.0/UDP 220.127.116.11:5060;branch=z9hG4bK1439e2a1;rport Max-Forwards: 70 From: "Anonymous" <sip:Anonymous@18.104.22.168>;;tag=as3bcef15a To: <sip:email@example.com:61482> Contact: <sip:Anonymous@22.214.171.124:5060> Call-ID: firstname.lastname@example.org:5060 CSeq: 102 INVITE User-Agent: voip.ms Date: Thu, 10 Aug 2017 22:30:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Anonymous" <sip:Anonymous@126.96.36.199>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 266
The bad news is this won’t be consumable by Unified Communications Manager in this present format. The good news, however, is that this can be fixed.
CUBE allows regular expression matching header messages as well as the ability to change that header messages. Often we see this use case when working with a carrier that doesn’t support a particular header or header format. In this case our own Cisco system doesn’t like the format so it’s game on.
To change a header you need to use a SIP profile.
To cut to the chase I used a SIP profile configuration to simply remove “Anonymous” and put a 10 digit Calling Party number in its place. In my case I used “3030000000”.
voice service voip
voice class sip-profiles 1
request ANY sip-header From modify "<sip:Restricted@" "<sip:3030000000@"
request ANY sip-header From modify "<sip:anonymous@" "<sip:3030000000@"
This profile then will need to be applied to a dial-peer somewhere in the call flow. I pick it up on the incoming dial peer with the “voice-class sip profiles” option.
dial-peer voice 1000 voip
description ***From CenturyLink**
voice-class sip profiles 1 inbound