我在通过运行在 Linux CENTOS 6.2 上的 ASTERISK PBX 拨打电话时遇到问题。
用例是从 /var/spool/asterisk/outbound/ 触发呼叫
呼叫者被拨打拨号计划执行：
Answer()
Wait(1.5)
Set(Timestamp=$<someformat)
Record(.../<filename>.wav,0,0,y)
HangUp()
我的 SIP 中继提供商是 nextiva。我从 wireshark 跟踪中注意到，在通话中断之前，nextiva 发送了一个 SIP：BYE 请求。
我攻击了wireshark跟踪以供参考：
536 110.28522 192.168.0.236 208.73.146.95 SIP/SDP Request: INVITE sip:[email protected], 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:[email protected]
540 110.491833 192.168.0.236 208.73.146.95 SIP/SDP Request: INVITE sip:[email protected], 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:[email protected]: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
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.
.
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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:[email protected]: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
有其他人遇到过类似的问题吗？
答案1
如果下游发送了 BYE，我总是向提供商开具票据并询问他们发送 BYE 的原因。通常，他们需要向另一家 ULC（底层运营商）开具票据才能解决问题。有时甚至来自更下游的地方。提供您的 CallID 和 PCAP，他们应该可以轻松找到它。