我正在尝试在Sipjs的帮助下为用户设置Asterisk语音聊天，并遵循Sipjs文档http://sipjs.com/guides/server-configuration/asterisk上给出的说明。用户被创建并连接。他们可以通过Zoiper互相打电话。但无法通过Sipjs或SipML5进行呼叫。当任何用户从Sipjs或SipMl5调用时。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'.
以及在安装DTLS证书期间，我得到"主机名:未知主机"。谁来指导我如何正确设置星号语音聊天
从我看到你的控制台-第一-扩展不存在于您的上下文中第二-在sip用户配置中设置"avpf=yes"在您的sip用户配置
中设置"encryption=yes"从你的标题我知道的是，你没有为它生成证书-
mkdir /etc/asterisk/keys
cd /usr/src/asterisk-13.1.0/contrib/scripts
/ast_tls_cert -C hostname -O "Some Name" -d /etc/asterisk/keys
下面是一个sip配置webphone的示例:
[1000]
host=dynamic
secret=YouPassword
context=from-internal
type=friend
encryption=yes
avpf=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
如果你仍然有问题，分享给我你的配置(希望你已经配置了webrtc支持)