Networking

linux (CENTOS 6.2) 上的 Asterisk PBX 掛斷 33 秒後正在錄製的呼叫

  • July 22, 2013

我通過在 linux CENTOS 6.2 上執行的 ASTERISK PBX 撥打電話時遇到問題。

案例是從 /var/spool/asterisk/outbound/ 觸發呼叫

呼叫者被撥打撥號計劃執行:

Answer()
Wait(1.5)
Set(Timestamp=$<someformat)
Record(.../<filename>.wav,0,0,y)
HangUp()

我的 SIP 中繼提供商是 nextiva。我從 Wireshark 跟踪中註意到的是,nextiva 就在呼叫被掛斷之前發送了一個 SIP: BYE 請求。

我攻擊了wireshark跟踪以供參考:

536 110.28522   192.168.0.236   208.73.146.95   SIP/SDP Request: INVITE sip:0116590224650@208.73.146.95, with session description
537 110.477662  208.73.146.95   192.168.0.236   SIP Status: 100 Trying
538 110.491041  208.73.146.95   192.168.0.236   SIP Status: 407 Proxy Authentication Required
539 110.491738  192.168.0.236   208.73.146.95   SIP Request: ACK sip:0116590224650@208.73.146.95
540 110.491833  192.168.0.236   208.73.146.95   SIP/SDP Request: INVITE sip:0116590224650@208.73.146.95, with session description
541 110.685694  208.73.146.95   192.168.0.236   SIP Status: 100 Trying
551 117.480397  208.73.146.95   192.168.0.236   SIP/SDP Status: 183 Session Progress, with session description
554 120.407182  208.73.146.95   192.168.0.236   SIP/SDP Status: 200 OK, with session description
555 120.407495  192.168.0.236   208.73.146.95   SIP Request: ACK sip:0116590224650@208.73.146.95:5060;transport=udp
556 121.40902   192.168.0.236   208.73.146.95   RTP PT=ITU-T G.711 PCMU, SSRC=0xE5D7E61, Seq=39878, Time=160 
557 121.429117  192.168.0.236   208.73.146.95   RTP PT=ITU-T G.711 PCMU, SSRC=0xE5D7E61, Seq=39879, Time=320 
558 SSRC=0x17D1D704, Seq=64350, Time=1164450752 
2152    151.356593  208.73.146.95   192.168.0.236   RTP PT=ITU-T G.711 PCMU, 
SSRC=0x17D1D704, Seq=64351, Time=1164450912 
.
.
.
.

2153    151.376572  208.73.146.95   192.168.0.236   RTP PT=ITU-T G.711 PCMU, SSRC=0x17D1D704, Seq=64352, Time=1164451072 
2156    151.409798  192.168.0.236   208.73.146.95   RTCP    Receiver Report   Source description   
2157    151.497917  208.73.146.95   192.168.0.236   SIP Request: BYE sip:706955271@192.168.0.236:5060
2158    151.498195  192.168.0.236   208.73.146.95   SIP Status: 200 OK
2164    152.125251  192.168.0.236   208.73.146.95   SIP Request: REGISTER 

其他人有熟悉的問題嗎?

在像這樣從下游發送 BYE 的情況下,我總是向我的提供商開一張票,並詢問他們為什麼發送 BYE。很多時候,他們需要使用另一家 ULC(基礎承運人)開票才能解決問題。有時它甚至比這更下游。提供您的 CallID 和 PCAP,他們應該可以跟踪它。

引用自:https://serverfault.com/questions/525356