<div dir="rtl"><div dir="ltr">Hello,</div><div dir="ltr"><br></div><div dir="ltr">  I am quite puzzled with this issue... A SIP phone calls an external SIP trunk via Freeswitch. In the "Ringing" packet leg B sends the <b>callee</b> name, but it is not passed to leg A. It is passed only after the OK is received. I have set ignore_display_updates to false, but it doesn't help. What else can I do?</div><div dir="ltr">I'm attaching the incoming and outgoing "Ringing" packets.</div><div dir="ltr"><br></div><div dir="ltr">                    Thanks! __Yehavi:</div><div dir="ltr"><br></div><div dir="ltr">recv 839 bytes from tls/[1.2.3.4]:5061 at 08:19:13.197453:<br>   ------------------------------------------------------------------------<br>   SIP/2.0 180 Ringing<br>   Via: SIP/2.0/TLS 1.2.3.4:6051;rport=35614;branch=z9hG4bK16U455UF7a0XH<br>   From: "Yehavi Bourvine" <<a href="mailto:sip%3A89444@1.2.3.4">sip:89444@1.2.3.4</a>>;tag=N9Krjm5v4DK4e<br>   To: <<a href="mailto:sip%3A85550@xxxxx.huji.ac.il">sip:85550@xxxxx.huji.ac.il</a>>;tag=N04936HrDQQpQ<br>   Call-ID: 54d1daf3-6a77-1238-dbbc-005056971340<br>   CSeq: 11047785 INVITE<br>   Contact: <sip:85550@1.2.3.4:5061;transport=tls><br>   User-Agent: FreeSWITCH-mod_sofia/1.8.7~64bit<br>   Accept: application/sdp<br>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br>   Supported: path, replaces<br>   Allow-Events: talk, hold, conference, refer<br>   Content-Length: 0<br>   X-FS-Display-Name: Yehavi Test Ofek<br>   X-FS-Display-Number: <a href="mailto:sip%3A85550@xxxxx.huji.ac.il">sip:85550@xxxxx.huji.ac.il</a><br>   X-FS-Support: update_display,send_info<br><span style="background-color:rgb(255,255,0)">   Remote-Party-ID: "Yehavi Test Ofek" <<a href="mailto:sip%3A85550@xxxx.huji.ac.il">sip:85550@xxxx.huji.ac.il</a>>;party=calling;privacy=off;screen=no</span><br>   <br></div><div dir="ltr">send 709 bytes to udp/[2.2.2.2]:5062 at 08:19:13.216944:<br>   ------------------------------------------------------------------------<br>   SIP/2.0 180 Ringing<br>   Via: SIP/2.0/UDP 2.2.2.2:5062;branch=z9hG4bK1547592<br>   From: "Yehavi Bourvine" <<a href="mailto:sip%3A89444@yyyy.huji.ac.il">sip:89444@yyyy.huji.ac.il</a>>;tag=1604186644<br>   To: <<a href="mailto:sip%3A85550@yyyyy.huji.ac.il">sip:85550@yyyyy.huji.ac.il</a>>;tag=2UHa7eB4086FS<br>   Call-ID: <a href="mailto:544809116@2.2.2.2">544809116@2.2.2.2</a><br>   CSeq: 2 INVITE<br>   Contact: <sip:85550@2.2.2.2:5060;transport=udp><br>   User-Agent: FreeSWITCH-mod_sofia/1.8.7~64bit<br>   Accept: application/sdp<br>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY<br>   Supported: path, replaces<br>   Allow-Events: talk, hold, conference, refer<br>   Content-Length: 0<br><span style="background-color:rgb(255,255,0)">   Remote-Party-ID: "85550" <<a href="mailto:sip%3A85550@yyyyy.huji.ac.il">sip:85550@yyyyy.huji.ac.il</a>>;party=calling;privacy=off;screen=no</span><br></div></div>