Hello,
thank you for helping.
I have the following scenario (TLS is enabled in SIP profile both Internal & External):
========================
caller incoming (udp 5080) to Server A (udp 5080). Server A should bridge call to Server B (over tls 5081).
========================
I am just testing out but seem not able because I am getting in the Invite showing is UDP and not TLS.
Please share with me how in Dialplan -> Inbound routes, I should set up to achieve it? Doable?
Thanks.
INVITE sip:68176331@sip.server.b.com:5061 SIP/2.0
Via: SIP/2.0/UDP sip.server.a.sg:5081;rport;branch=z9hG4bKFvSceyc2D84aF
Max-Forwards: 67
From: "69507009" <sip:69507009@sip.server.a.sg>;tag=0K0t3a1gN47Fr
To: <sip:68176331@sip.server.b.com:5061>
Call-ID: 07ec8633-73de-123d-9da1-9aac180d09ce
CSeq: 81933363 INVITE
Contact: <sip:mod_sofia@sip.server.a.sg:5081>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-FS-Support: update_display,send_info
P-Asserted-Identity: "69507009" <sip:69507009@sip.server.a.sg>
thank you for helping.
I have the following scenario (TLS is enabled in SIP profile both Internal & External):
========================
caller incoming (udp 5080) to Server A (udp 5080). Server A should bridge call to Server B (over tls 5081).
========================
I am just testing out but seem not able because I am getting in the Invite showing is UDP and not TLS.
Please share with me how in Dialplan -> Inbound routes, I should set up to achieve it? Doable?
Thanks.
INVITE sip:68176331@sip.server.b.com:5061 SIP/2.0
Via: SIP/2.0/UDP sip.server.a.sg:5081;rport;branch=z9hG4bKFvSceyc2D84aF
Max-Forwards: 67
From: "69507009" <sip:69507009@sip.server.a.sg>;tag=0K0t3a1gN47Fr
To: <sip:68176331@sip.server.b.com:5061>
Call-ID: 07ec8633-73de-123d-9da1-9aac180d09ce
CSeq: 81933363 INVITE
Contact: <sip:mod_sofia@sip.server.a.sg:5081>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 275
X-FS-Support: update_display,send_info
P-Asserted-Identity: "69507009" <sip:69507009@sip.server.a.sg>