Voip
未找到星號 + SIP 404
我想在家裡做一個小型的 Asterisk 伺服器。我在我的 Ubuntu 上安裝了星號，我使用 2 台電腦，以便相互連接。當我連接時，我可以在 Wireshark 中看到該註冊商正常。這是sip show peers命令的輸出：
Name/username Host Dyn Forcerport ACL Port Status uriel/uriel 192.168.1.101 D N 5060 Unmonitored vibrant/vibrant 192.168.1.100 D N 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]但問題是當我從 uriel 呼叫 sip:vibrant@192.168.1.200 時，我得到 404 not found。
對不起我的英語。
正如使用者 MealstroM 在這裡詢問的那樣，這是我的 sip.conf：
[vibrant] type=friend username=vibrant secret= host=dynamic context=tutorial nat=yes qualify=yes [uriel] type=friend username=uriel secret= host=dynamic context=tutorial nat=yes qualify=yes並且對於 cli sip 設置調試對等體充滿活力
uriel-desktop*CLI> sip set debug peer vibrant SIP Debugging Enabled for IP: 192.168.1.100 INVITE sip:uriel@192.168.1.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPj5uU12iuF3c7r4U7XBIgX36ORxjGapenJ Max-Forwards: 70 From: ;tag=v911t7.3Vk2K1-5Um-iWhFL6AmkL5uEq To: Contact: Call-ID: M1IPA30WrJWAmikvmIk1fikAEUSD4q5c CSeq: 4975 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 600 v=0 o=- 3535974498 3535974498 IN IP4 192.168.1.100 s=pjmedia c=IN IP4 192.168.1.100 t=0 0 a=X-nat:0 m=audio 40010 RTP/AVP 106 105 107 3 0 8 9 108 103 104 102 18 101 c=IN IP4 192.168.1.100 a=rtcp:40011 IN IP4 192.168.1.100 a=sendrecv a=rtpmap:106 speex/16000 a=rtpmap:105 speex/8000 a=rtpmap:107 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:108 AMR/8000 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:102 ILBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 25 lines) --- Sending to 192.168.1.100:5060 (no NAT) Using INVITE request as basis request - M1IPA30WrJWAmikvmIk1fikAEUSD4q5c Found peer 'vibrant' for '192.168.1.100' from 192.168.1.100:5060 Found RTP audio format 106 Found RTP audio format 105 Found RTP audio format 107 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 108 Found RTP audio format 103 Found RTP audio format 104 Found RTP audio format 102 Found RTP audio format 18 Found RTP audio format 101 Found audio description format speex for ID 106 Found audio description format speex for ID 105 Found audio description format speex for ID 107 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description format AMR for ID 108 Found audio description format ISAC for ID 103 Found audio description format ISAC for ID 104 Found audio description format ILBC for ID 102 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x20050170e (gsm|ulaw|alaw|g729|speex|speex16|ilbc|g722|h263p|mpeg4)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.100:40010 Looking for uriel in tutorial (domain 192.168.1.200) SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bKPj5uU12iuF3c7r4U7XBIgX36ORxjGapenJ;received=192.168.1.100;rport=5060 From: ;tag=v911t7.3Vk2K1-5Um-iWhFL6AmkL5uEq To: ;tag=as4078a435 Call-ID: M1IPA30WrJWAmikvmIk1fikAEUSD4q5c CSeq: 4975 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 [Jan 19 17:08:18] NOTICE[1081]: chan_sip.c:21614 handle_request_invite: Call from 'vibrant' to extension 'uriel' rejected because extension not found in context 'tutorial'. Scheduling destruction of SIP dialog 'M1IPA30WrJWAmikvmIk1fikAEUSD4q5c' in 6400 ms (Method: INVITE) ACK sip:uriel@192.168.1.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;rport;branch=z9hG4bKPj5uU12iuF3c7r4U7XBIgX36ORxjGapenJ Max-Forwards: 70 From: ;tag=v911t7.3Vk2K1-5Um-iWhFL6AmkL5uEq To: ;tag=as4078a435 Call-ID: M1IPA30WrJWAmikvmIk1fikAEUSD4q5c CSeq: 4975 ACK Content-Length: 0 --- (8 headers 0 lines) --- Really destroying SIP dialog 'M1IPA30WrJWAmikvmIk1fikAEUSD4q5c' Method: ACK Reliably Transmitting (NAT) to 192.168.1.100:5060: OPTIONS sip:vibrant@192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK41ab4d90;rport Max-Forwards: 70 From: "asterisk" ;tag=as47523e3d To: Contact: Call-ID: 1ef589e575f2263903cae0931bb27eb4@192.168.1.200:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Date: Thu, 19 Jan 2012 15:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;rport=5060;received=192.168.1.200;branch=z9hG4bK41ab4d90 Call-ID: 1ef589e575f2263903cae0931bb27eb4@192.168.1.200:5060 From: "asterisk" ;tag=as47523e3d To: ;tag=z9hG4bK41ab4d90 CSeq: 102 OPTIONS Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain Supported: replaces, 100rel, timer, norefersub Allow-Events: presence, message-summary, refer Content-Length: 0 --- (11 headers 0 lines) --- Really destroying SIP dialog '1ef589e575f2263903cae0931bb27eb4@192.168.1.200:5060' Method: OPTIONS現在是 extensions.conf
[tutorial] exten => uriel,1,Dial(SIP/uriel); exten => vibrant,2,Dial(SIP/vibrant);
[Jan 19 17:08:18] NOTICE[1081]: chan_sip.c:21614 handle_request_invite: Call from 'vibrant' to extension 'uriel' rejected because extension not found in context 'tutorial'.這就解釋了，您可以發布教程上下文的內容嗎？
解決了！將 extensions.conf 更改為：
[tutorial] exten => uriel,1,Dial(SIP/uriel@192.168.1.101); exten => vibrant,2,Dial(SIP/vibrant@192.168.1.200);我不知道這是否是正確的方法，但它有效