我有一个问题，我通过在Linux CENTOS 6.2上运行的ASTERISK PBX拨打电话。
用例是调用是从/ var / spool / asterisk / outbound /
拨打主叫方拨号scheme执行：
Answer() Wait(1.5) Set(Timestamp=$<someformat) Record(.../<filename>.wav,0,0,y) HangUp() 我的SIP干线提供者是nextiva。 我从Wireshark跟踪中注意到，nextiva在呼叫被删除之前发送一个SIP：BYE请求。
我攻击wireshark跟踪作为参考：
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 . . . . 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
有没有其他人有熟悉的问题？
在BYE从下游发送的情况下，我总是与供应商打开一张票，询问为什么他们发送了BYE。 通常情况下，这是另一个ULC（底层运营商），他们需要打开票来解决它。 有时甚至比这更下游。 提供您的CallID和PCAPs，这不应该成为他们追踪的问题。