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社区首页 >问答首页 >无法使用jssip库连接到我的Asterisk服务器，密码错误

问无法使用jssip库连接到我的Asterisk服务器，密码错误
EN

Stack Overflow用户
提问于 2017-11-11 03:25:08
回答 2查看 2.6K关注 0票数 0

我可以使用jssip连接到我的freePbx服务器。但是在Asterisk日志中使用错误的密码注册失败。我可以使用相同的PjSip扩展密码连接和注册none WebRtc和WebSocket客户端。

它在.net库Ozeki中使用相同的用户、pass等进行工作，但在nodejs和webRtc中不起作用。

星号日志：

代码语言：javascript
复制
ERROR[24788]: res_http_websocket.c:506 ws_safe_read: Error readng from webScoket: Connection rest by peer.
NOTICE[33279]: chan_sip.c28486 handle_request_register: Registeration from '<sip:170@192.168.0.210>' faild for 192.168.0.250:43751 - wrong password

jssip conf：

代码语言：javascript
复制
var socket = new JsSIP.WebSocketInterface('ws://192.168.0.210:8089/ws');

var configuration = {
    sockets           : [ socket ],
    authorization_user: '170',
    uri               : 'sip:170@192.168.0.210',
    ws_servers        : 'ws://192.168.0.210:8089/ws',
    password          : '856589',
    realm             : '192.168.0.210',
    display_name      : '170',
    contact_uri       : 'sip:170@192.168.0.210'
};

jssip日志：

代码语言：javascript
复制
olden@golden-client:~/LocalFiles/projects/jssip$ node ./index.js  JsSIP version 3.0.21 +0ms  jssip-node-websocket new() [url:"ws://192.168.0.210:8088/ws", options:undefined] +0ms
  JsSIP:UA new() [configuration:{ sockets: [ NodeWebSocket { _url: 'ws://192.168.0.210:8088/ws', _options: {}, _sipUri: 'sip:192.168.0.210:8088;transport=ws', _viaTransport: 'WS', _ws: null } ], authorization_user: '170', uri: 'sip:170@192.168.0.210:5060', ws_servers: 'ws://192.168.0.210:8088/ws', password: '170170', realm: '192.168.0.210', display_name: '170', contact_uri: 'sip...@192.168.0.210' }] +0ms
  JsSIP:Transport new() +0ms
  JsSIP:UA configuration parameters after validation: +15ms
  JsSIP:UA - authorization_user: "170" +0ms
  JsSIP:UA - password: NOT SHOWN +0ms
  JsSIP:UA - realm: "192.168.0.210" +0ms
  JsSIP:UA - ha1: NOT SHOWN +0ms
  JsSIP:UA - display_name: "170" +0ms
  JsSIP:UA - uri: sip:170@192.168.0.210:5060 +0ms
  JsSIP:UA - contact_uri: {"_parameters":{},"_headers":{},"_scheme":"sip","_user":"170","_host":"192.168.0.210"} +0ms
  JsSIP:UA - instance_id: "f3a58a94-b426-4a8d-8b15-9f7208a42f9b" +0ms
  JsSIP:UA - use_preloaded_route: false +0ms
  JsSIP:UA - session_timers: true +0ms
  JsSIP:UA - no_answer_timeout: 60000 +0ms
  JsSIP:UA - register: true +0ms
  JsSIP:UA - register_expires: 600 +0ms
  JsSIP:UA - registrar_server: sip:192.168.0.210:5060 +0ms
  JsSIP:UA - connection_recovery_max_interval: null +0ms
  JsSIP:UA - connection_recovery_min_interval: null +0ms
  JsSIP:UA - via_host: "192.168.0.210" +0ms
  JsSIP:UA start() +0ms
  JsSIP:Transport connect() +1ms
connecting
  jssip-node-websocket connect() +18ms
  jssip-node-websocket WebSocket connecting [url:"ws://192.168.0.210:8088/ws"] +0ms
  jssip-node-websocket WebSocket open [url:"ws://192.168.0.210:8088/ws"] +18ms
connected
  JsSIP:Transport send() +22ms
  JsSIP:Transport sending message:
  JsSIP:Transport
  JsSIP:Transport REGISTER sip:192.168.0.210:5060 SIP/2.0
  JsSIP:Transport Via: SIP/2.0/WS 192.168.0.210;branch=z9hG4bK160956
  JsSIP:Transport Max-Forwards: 69
  JsSIP:Transport To: <sip:170@192.168.0.210:5060>
  JsSIP:Transport From: "170" <sip:170@192.168.0.210:5060>;tag=s9bkng5emf
  JsSIP:Transport Call-ID: 6ra4f68tegdmqllfoa7kja
  JsSIP:Transport CSeq: 1 REGISTER
  JsSIP:Transport Contact: <sip...@192.168.0.210>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:f3a58a94-b426-4a8d-8b15-9f7208a42f9b>";expires=600
  JsSIP:Transport Expires: 600
  JsSIP:Transport Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
  JsSIP:Transport Supported: path,gruu,outbound
  JsSIP:Transport User-Agent: JsSIP 3.0.21
  JsSIP:Transport Content-Length: 0
  JsSIP:Transport
  JsSIP:Transport
  JsSIP:Transport  +0ms
  jssip-node-websocket send() +3ms
  jssip-node-websocket WebSocket message received +3ms
  JsSIP:Transport received text message:
  JsSIP:Transport
  JsSIP:Transport SIP/2.0 401 Unauthorized
  JsSIP:Transport Via: SIP/2.0/WS 192.168.0.210;branch=z9hG4bK160956;received=192.168.0.3;rport=35832
  JsSIP:Transport From: "170" <sip:170@192.168.0.210:5060>;tag=s9bkng5emf
  JsSIP:Transport To: <sip:170@192.168.0.210:5060>;tag=as0f727a9b
  JsSIP:Transport Call-ID: 6ra4f68tegdmqllfoa7kja
  JsSIP:Transport CSeq: 1 REGISTER
  JsSIP:Transport Server: FPBX-13.0.192.19(13.12.1)
  JsSIP:Transport Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  JsSIP:Transport Supported: replaces, timer
  JsSIP:Transport WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="078cc3e4"
  JsSIP:Transport Content-Length: 0
  JsSIP:Transport
  JsSIP:Transport
  JsSIP:Transport  +3ms
  JsSIP:DigestAuthentication authenticate() | response generated +0ms
  JsSIP:Transport send() +10ms
  JsSIP:Transport sending message:
  JsSIP:Transport
  JsSIP:Transport REGISTER sip:192.168.0.210:5060 SIP/2.0
  JsSIP:Transport Via: SIP/2.0/WS 192.168.0.210;branch=z9hG4bK4224730
  JsSIP:Transport Max-Forwards: 69
  JsSIP:Transport To: <sip:170@192.168.0.210:5060>
  JsSIP:Transport From: "170" <sip:170@192.168.0.210:5060>;tag=s9bkng5emf
  JsSIP:Transport Call-ID: 6ra4f68tegdmqllfoa7kja
  JsSIP:Transport CSeq: 2 REGISTER
  JsSIP:Transport Authorization: Digest algorithm=MD5, username="170", realm="asterisk", nonce="078cc3e4", uri="sip:192.168.0.210:5060", response="26576b92bf68c3d315d480b02d53783c"
  JsSIP:Transport Contact: <sip...@192.168.0.210>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:f3a58a94-b426-4a8d-8b15-9f7208a42f9b>";expires=600
  JsSIP:Transport Expires: 600
  JsSIP:Transport Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
  JsSIP:Transport Supported: path,gruu,outbound
  JsSIP:Transport User-Agent: JsSIP 3.0.21
  JsSIP:Transport Content-Length: 0
  JsSIP:Transport
  JsSIP:Transport
  JsSIP:Transport  +0ms
  jssip-node-websocket send() +11ms
  jssip-node-websocket WebSocket message received +1ms
  JsSIP:Transport received text message:
  JsSIP:Transport
  JsSIP:Transport SIP/2.0 403 Forbidden
  JsSIP:Transport Via: SIP/2.0/WS 192.168.0.210;branch=z9hG4bK4224730;received=192.168.0.3;rport=35832
  JsSIP:Transport From: "170" <sip:170@192.168.0.210:5060>;tag=s9bkng5emf
  JsSIP:Transport To: <sip:170@192.168.0.210:5060>;tag=as0f727a9b
  JsSIP:Transport Call-ID: 6ra4f68tegdmqllfoa7kja
  JsSIP:Transport CSeq: 2 REGISTER
  JsSIP:Transport Server: FPBX-13.0.192.19(13.12.1)
  JsSIP:Transport Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  JsSIP:Transport Supported: replaces, timer
  JsSIP:Transport Content-Length: 0
  JsSIP:Transport
  JsSIP:Transport
  JsSIP:Transport  +3ms

{ response:
   IncomingResponse {
     data: 'SIP/2.0 403 Forbidden\r\nVia: SIP/2.0/WS 192.168.0.210;branch=z9hG4bK4224730;received=192.168.0.3;rport=35832\r\nFrom: "170" <sip:170@192.168.0.210:5060>;tag=s9bkng5emf\r\nTo: <sip:170@192.168.0.210:5060>;tag=as0f727a9b\r\nCall-ID: 6ra4f68tegdmqllfoa7kja\r\nCSeq: 2 REGISTER\r\nServer: FPBX-13.0.192.19(13.12.1)\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\nSupported:replaces, timer\r\nContent-Length: 0\r\n\r\n',
     headers:
      { Via: [Array],
        From: [Array],
        To: [Array],
        'Call-ID': [Array],
        CSeq: [Array],
        Server: [Array],
        Allow: [Array],
        Supported: [Array],
        'Content-Length': [Array] },
     method: 'REGISTER',
     via:
      { protocol: 'SIP',
        transport: 'WS',
        host_type: 'IPv4',
        host: '192.168.0.210',
        branch: 'z9hG4bK4224730',
        received: '192.168.0.3' },
     via_branch: 'z9hG4bK4224730',
     call_id: '6ra4f68tegdmqllfoa7kja',
     cseq: 2,
     from: NameAddrHeader { _uri: [Object], _parameters: [Object], _display_name: '170' },
     from_tag: 's9bkng5emf',
     to:
      NameAddrHeader {
        _uri: [Object],
        _parameters: [Object],
        _display_name: undefined },
     to_tag: 'as0f727a9b',
     body: '',
     sdp: null,
     status_code: 403,
     reason_phrase: 'Forbidden' },
  cause: 'Rejected' }
freepbx
jssip
asterisk
sip
voip
EN

回答 2

Stack Overflow用户

发布于 2017-11-18 19:19:51

正如预期的那样，注册过程遵循RFC3261中指定的过程，也就是说，第一个寄存器由401错误响应应答，该响应导致第二个寄存器具有授权信息。在这种情况下，服务器会拒绝这些数据。

代码语言：javascript
复制
170 (REGISTER) ---------------------> Asterisk
170 <----------(401 - Not authorized) Asterisk
170 (REGISTER with Auth info) ------> Asterisk
170 <----------------(407 - Rejected) Asterisk

因此，asterisk拒绝注册，因为它不喜欢第二个register消息。

代码语言：javascript
复制
Authorization: Digest algorithm=MD5, username="170", realm="asterisk", nonce="078cc3e4", uri="sip:192.168.0.210:5060", response="26576b92bf68c3d315d480b02d53783c"

领域“星号”与jssip中的配置不同，它也不遵循RFC中指定的内容：

将对收到的请求进行身份验证的用户代理或代理服务器的

操作员必须遵守以下为其服务器创建领域字符串的指导原则：

  • 领域字符串必须全局唯一。建议领域字符串包含主机名或域名，遵循RFC2617的第3.2.1节中的建议。

  • 领域字符串应提供可呈现给用户的人类可读标识符。

因此，您应该将sip.conf配置为在星号中定义领域字符串，以遵循上述规则。无论如何，请注意，即使jssip配置了不同的“领域”，在第二个寄存器中也会使用第一个响应中的字段。所以，我不认为这是问题所在。

另一个是不包括用户名的第二个寄存器中的uri="sip:192.168.0.210:5060“。如果它遵循下一个示例，取自"sip:170:192.168.0.210:5060“

代码语言：javascript
复制
Authorization: Digest username="bob",..., uri="sip:bob@biloxi.com",..."

我不明白为什么uri不包含在jssip.conf中指定的用户名。我将从配置中删除"authorization_user“，因为它是一个可选参数和测试。

票数 2
EN

Stack Overflow用户

发布于 2020-08-18 06:22:55

禁用chan_sip (设置>高级设置-拨号计划和操作- SIP通道Driver=pjsip)。然后它就成功了！

票数 0
EN
页面原文内容由Stack Overflow提供。腾讯云小微IT领域专用引擎提供翻译支持
原文链接：

https://stackoverflow.com/questions/47229835

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