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不知道是否應該問我這個問題。Asterisk在掛斷時返回481
我正在開發一個簡單的voip應用程序。我可以打給其他用戶，但不能掛斷電話。當客戶端發送BYE請求時，服務器回答481 - 呼叫段交易不存在。 下面是客戶端日誌：
INVITE sip:[email protected]:51110;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:5060;branch=z9hG4bK06952c7a;rport
Max-Forwards: 70
From: "First" <sip:[email protected]>;tag=as746cc61d
To: <sip:[email protected]:51110;transport=UDP>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Date: Tue, 07 Mar 2017 11:52:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "First" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 310
v=0
o=root 2015785808 2015785808 IN IP4 172.20.4.7
s=Asterisk PBX 11.16.0
c=IN IP4 172.20.4.7
t=0 0
m=audio 13952 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKnCqg
Contact: <sip:[email protected]:51110;transport=UDP>
To: <sip:[email protected];transport=UDP>;tag=YU2R
From: <sip:[email protected];transport=UDP>;tag=as746cc61d
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: presence, kpml, talk
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bK06952c7a;rport
Contact: <sip:[email protected]:51110;transport=UDP>
To: <sip:[email protected];transport=UDP>;tag=YU2R
From: "First" <sip:[email protected];transport=>;tag=as746cc61d
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Accept: application/sdp, application/sdp
Accept-Language: en
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
Allow-Events: presence, kpml, talk
Content-Length: 247
v=0
o=Z 13 13 IN IP4 95.86.129.80
s=Test
c=IN IP4 95.86.129.80
t=0 0
m=audio 50000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode = 30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
ACK sip:[email protected]:51110;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:5060;branch=z9hG4bK54ba94a0;rport
Max-Forwards: 70
From: "First" <sip:[email protected]>;tag=as746cc61d
To: <sip:[email protected]:51110;transport=UDP>;tag=YU2R
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Content-Length: 0
BYE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKAZsQ
Max-Forwards: 70
From: <sip:[email protected];transport=UDP>;tag=as746cc61d
To: <sip:[email protected];transport=UDP>;tag=YU2R
Contact: <sip:[email protected]:51110;transport=UDP>
CSeq: 2 BYE
User-Agent: TestSoftphone
Call-ID: [email protected]:5060
Content-Length: 0
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKAZsQ;received=172.20.1.40;rport=51110
From: <sip:[email protected];transport=UDP>;tag=as746cc61d
To: <sip:[email protected];transport=UDP>;tag=YU2R
Call-ID: [email protected]:5060
CSeq: 2 BYE
Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
我讀，如果To標籤或From標籤或Call-ID是對話外，但看起來像標籤和IDS都ok 481可能發生。我沒有收到To的標籤，所以我生成Ringing數據包，其中To標籤已設置，這可能是問題所在。
編輯。我在sip_custom.conf中添加了「pedantic = no」，現在它可以工作。雖然我不知道這個對話框有什麼問題。
不清楚什麼答案481？它是星號嗎？基於kamailio的軟交換？什麼？這是openser系列的錯誤配置（kamailio，opensips）的常見錯誤。您應該始終在源代碼中包含源IP，否則很難閱讀，而且沒有人會這樣做。 – arheops
是的，它是星號，日誌來自我的客戶。我編輯了這個問題。 Zoiper（SIP軟電話）可以正常使用此星號服務器，但發送BYE請求時遇到問題。答案481是日誌中的最後一個答案，正如我理解的那樣，服務器通知我它不知道該呼叫，即我用ID標記了一些東西。 – Jamil
這意味着它無法識別該BYE的對話框。或者，也許已經發送BYE到其他IP。問題需要調試，沒辦法回答這個問題。調試在SO上是不重要的。 – arheops