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