<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">It may be possible with code to just always add a contact, not sure there is any reason not to, just isn’t typically necessary so it wasn’t there.<br class=""><div><br class=""><blockquote type="cite" class=""><div class="">On Sep 12, 2018, at 8:17 AM, Sebastian Zenzerović <<a href="mailto:sebastian.zenzerovic@inet.hr" class="">sebastian.zenzerovic@inet.hr</a>> wrote:</div><br class="Apple-interchange-newline"><div class="">
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<div class="moz-text-flowed" style="font-family: -moz-fixed;
font-size: 14px;" lang="x-unicode">Hello,
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is it possible to add (configuration option/variable?) Contact
header to outgoing gateway ping messages sent by Freeswitch?
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Remote SIP peer is answering "403 forbidden" as there is no
Contact header present in SIP OPTIONS.
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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 (<a class="moz-txt-link-freetext" href="https://aka.ms/dr/">https://aka.ms/dr/</a>).
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Here are the requirements:
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• Must: When placing calls to the Direct Routing interface, the
'CONTACT' header must have the SBC FQDN in the URI hostname
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• Syntax: Contact: <phone number>@<FQDN of the
SBC>:<SBC Port>;<transport type>
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• If the parameter is not configured correctly, OPTIONS are
rejected with a '403 Forbidden' message
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Freeswitch log:
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OPTIONS <a class="moz-txt-link-freetext" href="sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls">sip:sip-du-a-euno.pstnhub.microsoft.com;transport=tls</a>
SIP/2.0
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Via: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS
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Max-Forwards: 70
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From: <a class="moz-txt-link-rfc2396E" href="sip:sbc.xxx.xx"><sip:sbc.xxx.xx></a>;tag=0aB5jm3SF07ve
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To: <a class="moz-txt-link-rfc2396E" href="sip:sbc.xxx.xx"><sip:sbc.xxx.xx></a>
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Call-ID: 101b723c-3113-1237-6092-00155d000d46
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CSeq: 128016309 OPTIONS
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User-Agent: FreeSWITCH-mod_sofia/1.8.1-2-4f54cff36a~64bit
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Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY
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Supported: timer, path, replaces
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Allow-Events: talk, hold, conference, refer
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Content-Length: 0
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SIP/2.0 403 Forbidden
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FROM: <a class="moz-txt-link-rfc2396E" href="sip:sbc.xxx.xx"><sip:sbc.xxx.xx></a>;tag=0aB5jm3SF07ve
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TO: <a class="moz-txt-link-rfc2396E" href="sip:sbc.xxx.xx"><sip:sbc.xxx.xx></a>
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CSEQ: 128016309 OPTIONS
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CALL-ID: 101b723c-3113-1237-6092-00155d000d46
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VIA: SIP/2.0/TLS x.x.x.x;branch=z9hG4bKvjX0rU7K5QtKS
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REASON:
Q.850;cause=21;text="4da4bb47-5ace-41b2-ba4b-6516fb4771c2;Record-Route
and Contact headers are missing"
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CONTENT-LENGTH: 0
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ALLOW: INVITE
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ALLOW: ACK
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ALLOW: OPTIONS
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ALLOW: CANCEL
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ALLOW: BYE
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ALLOW: NOTIFY
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SERVER: Microsoft.PSTNHub.SIPProxy v.2018.9.10.4 i.EUNO.4
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