java dtls server_尝试使用sipjs在网站用户之间设置Asterisk进行语音聊天 . 但无法配置DTLS证书 . 得到“主机名：未知主机”...

我正在尝试使用Sipjs的帮助为用户设置Asterisk语音聊天，遵循SIPJS docs http://sipjs.com/guides/server-configuration/asterisk上给出的说明 . 用户已创建并已连接 . 他们可以通过Zoiper互相打电话 . 但无法通过Sipjs或SipML5调用 . 当任何人用户从Sipjs或SipMl5进行呼叫时 . 控制台显示以下错误：

Connected to Asterisk 11.20.0 currently running on asterix (pid = 13719)

[Oct 14 05:25:22] NOTICE[13735][C-00000000]: chan_sip.c:25844 handle_request_invite: Call from '' (88.150.240.102:5071) to extension '90041215085741' rejected because extension not found in context 'default'.

[Oct 14 05:25:46] NOTICE[13735][C-00000001]: chan_sip.c:10005 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 23496 UDP/TLS/RTP/SAVPF 109 9 0 8

[Oct 14 05:25:46] WARNING[13735][C-00000001]: chan_sip.c:10398 process_sdp: Rejecting secure audio stream without encryption details: audio 23496 UDP/TLS/RTP/SAVPF 109 9 0 8

[Oct 14 05:25:54] WARNING[13735]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 1faf349623b90d4f62fe562ae66d6c45 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[Oct 14 05:25:55] NOTICE[13735][C-00000002]: chan_sip.c:25844 handle_request_invite: Call from '' (88.150.240.102:5070) to extension '0041215085741' rejected because extension not found in context 'default'.

并且在安装DTLS证书期间，我得到“主机名：未知主机” . 有人请指导我如何正确设置Asterisk语音聊天 .