Skype for Business

XO/Sonus and SIM Ring

I recently spent countless hours trying to deal with XO and getting SIM ring to work with Skype for Business. After several suggestions from support, all of which said something different and all of which where wrong. I finally got high enough in the support chain for somebody to give me an example header of one that worked. All of the other reps refused to help me out.

XO is not E.164 compliant either, they only want the 10 digits (and they also only send 10 digits), so make sure to catch that in your SBC.

When doing a sim ring or forward, the diversion header should have the XO number and the From can have any number (I had a lot of different variations on this from XO)

The configuration involves three settings you need to make

  1. In Skype for Business control panel for the Trunk I had to enable the checkbox for Forward History. DO NOT check enable forward P-Asserted-Identity data, it will cause a call loop
  2. At the sonus.
    1. On the signaling group to XO set the Play Ringback to always
    2. I had to create thee additional Redirecting Number translations, all optional . Make sure that these are on the Transformation table that are assigned from the SFB Mediation server (Check Call Routing Table, then the one that is from SfB find the Transformation table assigned and edit that one)
Input Field Type Input Field Value Output Field Type Output Field Value
Redirecting Number <BLANK> Redirecting Number <BLANK>
Redirecting Number (.*) Redirecting Number \1
Redirecting Number \+1(.*) Redirecting Number \1

What this will return you is a header that looks like the following




Via: SIP/2.0/UDP;branch=z9hG4bK02B23c13b7effe83a7a

From: “<NAME>” <sip:9999999777@;pstn-params=9084818088>;tag=gK022003c0

To: <sip:<SOMENUMBER>>

Call-ID: 1795319449_95961084@

CSeq: 15027 INVITE

Max-Forwards: 69


Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed

Contact: “sipp” <sip:9999999777@>

Diversion: <sip:<XONumber>>;privacy=off;screen=no; reason=no-answer; counter=1

Supported: timer,100rel

Session-Expires: 1800

Min-SE: 90

Content-Length:  180

Content-Disposition: session; handling=required

Content-Type: application/sdp



o=Sonus_UAC 30005 27570 IN IP4

s=SIP Media Capabilities

c=IN IP4

t=0 0

m=audio 27820 RTP/AVP 0

a=rtpmap:0 PCMU/8000






So break this down:

  • The From address will now be the original number of the calling party
  • The To will be the number setup for sim ring
  • The Number in the Diversion Header will be the caller number (the number being dialing)
  • The Invite will be the number in the SIM Ring setup

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