v20: “Unidentified Incoming Call” (Event 30051) on Orange SIP trunk, INVITE uses tel: URI, no Inbound Parameters tab to adjust source identification

[ronald.vlavonou@pdigitalc]

Hello,
I’m having an issue with inbound DID calls on an Orange SIP trunk (Côte d’Ivoire on 3CX Version 20.0 Update 10 (Build 1621 - AI 1.6.57),Linux.
Inbound calls to our DIDs fail. The 3CX Event Log shows this warning:

Event ID 30051 – “Unidentified Incoming Call. Review INVITE and adjust source identification”
The INVITE received (numbers partially masked):

INVITE tel:+2252025XXXXX SIP/2.0
Via: SIP/2.0/UDP 10.203.5.137:5060;branch=...
Record-Route: <sip:10.203.5.137:5060;lr;...>
Contact: <sip:10.203.5.137:5060;...>
To: <sip:[email protected]:5060;transport=udp;user=phone>
From: 01711XXXXX <tel:01711XXXXX;noa=subscriber;srvattri=national;phone-context=+225>;tag=...
Call-ID: [email protected]
P-Asserted-Identity: <tel:01711XXXXX>
P-Called-Party-ID: <tel:+2252025XXXXX>
Content-Type: application/sdp
Supported: timer, 100rel, histinfo, early-session

What I notice:

• The Request-URI is a tel: URI (INVITE tel:+2252025XXXXX), not sip:user@host.
• The dialed number is also in To (with user=phone) and in P-Called-Party-ID.
• There is no rinstance parameter, and the source is the operator’s SBC (10.203.5.137).

The DIDs are declared in the trunk’s DIDs tab [in which format: +2252025XXXXX but the call is never matched to the trunk.

In 3CX v18, I could change the field used to identify the DID in the trunk’s Inbound Parameters tab. In v20 I can’t find this tab or any equivalent option.


Questions:

1. In v20, how is the called number / trunk identified for a tel: Request-URI? Is there any setting left to choose the SIP field (Request-URI, To, P-Called-Party-ID)?
2. Is the trunk identified by source IP when no registration is used? If yes, where is this configured in v20?
3. Is this a known limitation or bug with tel: URIs in v20, and is there a recommended workaround (other than asking the operator to send a sip: URI)?

 

[Alejandro_3CX]

Hi @ronald.vlavonou@pdigitalc you can edit the template from the advanced menu ---- templates ---- voip provider templates.

Please follow the documentation outlined in this link

Once done you have edited your inbound parameters, go to the voice and chat menu ---- select the sip trunk and click on switch template, there you can select the new custom template.

Have a good day!