Removing video codecs from SIP profile did solve the issue. Thanks Valli. -- Just for reference, here is the offer to that Panasonic phone before the fix: )Hhe.E@C¸ª%æÙÄÄs3ºINVITE <mailto:Hhe.E@C¸ª%25æÙÄÄs3ºINVITE> sip:Account1@1.1.217.127:5060;received=1.1.217.127:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.131.37;rport;branch=z9hG4bKttaHKmXNgBtSQ Max-Forwards: 67 From: "Debug Proxy" <sip:+15145552353@1.1.131.37>;tag=rDQF79t14QmKr To: <sip:9919605@dev-proxy.xxx.com> Call-ID: cbecb6ef-4300-1239-dfa0-000c29124868 CSeq: 22952030 INVITE Contact: <sip:mod_sofia@1.1.131.37:5060> User-Agent: Acme CloudPBX Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, 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 Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 397 X-DNIS: 9919605 Remote-Party-ID: "Debug Proxy" <sip:+15145552353@1.1.131.37>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1594981106 1594981107 IN IP4 1.1.131.37 s=FreeSWITCH c=IN IP4 1.1.131.37 t=0 0 m=audio 30538 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 25520 RTP/AVP 34 b=AS:1024 a=rtpmap:34 H263/90000 a=rtcp-fb:34 ccm fir a=rtcp-fb:34 ccm tmmbr a=rtcp-fb:34 nack