Hello, is it possible to add (configuration option/variable?) Contact header to outgoing gateway ping messages sent by Freeswitch? Remote SIP peer is answering "403 forbidden" as there is no Contact header present in SIP OPTIONS. Not a big deal since FS sees the peer as available/UP anyway but could be a interoperability enhancement between FS and Microsoft Direct routing service (https://aka.ms/dr/). Here are the requirements: • Must: When placing calls to the Direct Routing interface, the 'CONTACT' header must have the SBC FQDN in the URI hostname • Syntax: Contact: <phone number>@<FQDN of the SBC>:<SBC Port>;<transport type> • If the parameter is not configured correctly, OPTIONS are rejected with a '403 Forbidden' message Freeswitch log: OPTIONS sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS Max-Forwards: 70 From: <sip:sbc.xxx.xx>;tag=0aB5jm3SF07ve To: <sip:sbc.xxx.xx> Call-ID: 101b723c-3113-1237-6092-00155d000d46 CSeq: 128016309 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 SIP/2.0 403 Forbidden FROM: <sip:sbc.xxx.xx>;tag=0aB5jm3SF07ve TO: <sip:sbc.xxx.xx> CSEQ: 128016309 OPTIONS CALL-ID: 101b723c-3113-1237-6092-00155d000d46 VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS REASON: Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route and Contact headers are missing" CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: ACK ALLOW: OPTIONS ALLOW: CANCEL ALLOW: BYE ALLOW: NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4 Thanks. BR, Sebastian