Нет ответа 200(ОК)

Форум для обсуждения KTS 4м GSM VoIP шлюза
Правила форума
Пожалуйста, будьте уважительны к участникам форума.
Avantelecom
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Зарегистрирован: 08 сен 2017, 13:50

Нет ответа 200(ОК)

Сообщение Avantelecom »

Добрый день.Шлюз s/n: KTS4MS640098i ver: 4.1.5.

Имеется проблема на всех 4 портах. При исходящих вызовах, когда удаленная сторона поднимает трубку с ее стороны не приходит ответа 200(ОК) - вообще ничего не приходит, просто тишина.
Проблема плавающая, появляется в разных портах случайным образом, лечится перезагрузкой шлюза.
kts
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Re: Нет ответа 200(ОК)

Сообщение kts »

Добрый день.
На серии MS была такая проблема, но мы ее решили в версии 4.1.5
Сегодня еще раз перепроверим.
Avantelecom
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Зарегистрирован: 08 сен 2017, 13:50

Re: Нет ответа 200(ОК)

Сообщение Avantelecom »

Логи собирать надо?
Avantelecom
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Зарегистрирован: 08 сен 2017, 13:50

Re: Нет ответа 200(ОК)

Сообщение Avantelecom »

Потому что сегодня опять возникла эта проблема
kts
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Re: Нет ответа 200(ОК)

Сообщение kts »

Добрый день.
У нас не обнаруживается похожая проблема.
Прошивка версии 4.1.5
Вот кусок лога прохождения звонка

2017-09-12 15:49:57 User.Notice 192.168.1.100 Dec 1 20:30:54 kts_gate: [truncated] SIP: 20:30:54.353 pjsua_core.c RX 1115 bytes Request msg INVITE/cseq=102 (rdata0x444e3c) from UDP 192.168.1.1:5060:<010>INVITE sip:8xxxxxxxxx@192.168.1.100:5061 SIP/2.0<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK580e3c38;rport<013><010>Max-Forwards: 70<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061><013><010>Contact: <sip:300@192.168.1.1><013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 102 INVITE<013><010>User-Agent: Asterisk PBX 1.6.1.17<013><010>Date: Tue, 12 Sep 2017 12:49:54 GMT<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<013><010>Supported: replaces, timer<013><010>Content-Type: application/sdp<013><010>Content-Length: 557<013><010><013><010>v=0<013><010>o=root 517671258 517671258 IN IP4 192.168.1.1<013><010>s=Asterisk PBX 1.6.1.17<013><010>c=IN IP4 192.168.1.1<013><010>t=0 0<013><010>m=audio 19948 RTP/AVP 0 4 3 8 112 5 10 7 18 2 9 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:4 G723/8000<013><010>a=fmtp:4 annexa=no<013><010>a=rtpmap:3 GSM/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:112 AAL2-G726-32/8000<013><010>a=rtp
2017-09-12 15:49:57 User.Debug 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: CheckAllowedOnModem: '8xxxxxxxxx'
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Call remote NAT type is 0 (Unknown)
2017-09-12 15:49:57 User.Notice 192.168.1.100 Dec 1 20:30:54 kts_gate: SIP: 20:30:54.417 pjsua_core.c TX 297 bytes Response msg 100/INVITE/cseq=102 (tdta0x45a178) to UDP 192.168.1.1:5060:<010>SIP/2.0 100 Trying<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK580e3c38<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061><013><010>CSeq: 102 INVITE<013><010>Content-Length: 0<013><010><013><010><010>--end msg--
2017-09-12 15:49:57 User.Debug 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Incoming call From: "300" <sip:300@192.168.1.1> To: <sip:8xxxxxxxxx@192.168.1.100:5061>
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Dialing number: 8xxxxxxxxx
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: AddCommand: ATD8xxxxxxxxx; (imm: 0 q: 0)
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: WriteMuxCmd(0): ATD8xxxxxxxxx;
2017-09-12 15:49:57 User.Debug 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Send SESSION PROGRESS
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Call remote NAT type is 0 (Unknown)
2017-09-12 15:49:57 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Attach media.
2017-09-12 15:49:58 User.Debug 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Set Volume (Rx: 1)(Tx: 1)
2017-09-12 15:49:58 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: CODEC: PCMU, payload: 0
2017-09-12 15:49:58 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Codec: PCMU ptime: 20. (Vinetic ID: 8).
2017-09-12 15:49:58 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Media updates, stream #1 PCMU (sendrecv)
2017-09-12 15:49:58 User.Info 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: Media for Call is active
kts
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Зарегистрирован: 18 янв 2008, 13:36

Re: Нет ответа 200(ОК)

Сообщение kts »

192.168.1.1:5060:<010>SIP/2.0 183 Session Progress<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK580e3c38<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>CSeq: 102 INVITE<013><010>Contact: <sip:0001@192.168.1.100:5061;ob><013><010>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<013><010>Content-Type: application/sdp<013><010>Content-Length: 290<013><010><013><010>v=0<013><010>o=root 944069454 944069455 IN IP4 192.168.1.100<013><010>s=ktsgate<013><010>c=IN IP4 192.168.1.100<013><010>t=0 0<013><010>a=X-nat:0 Unknown<013><010>m=audio 4000 RTP/AVP 0 101<013><010>a=rtcp:4001 IN IP4 192.168.1.100<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=ptime:20<013><010>a=maxptime:60<013><010>a=sendrecv<013><010><010>--end msg--
2017-09-12 15:49:58 User.Debug 192.168.1.100 Dec 1 20:30:54 kts_gate: 01: on_call_state finish
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: PR[0]: +COLP: "8xxxxxxxxx",129,"",0,""
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: PR[1]: +COLP: "8xxxxxxxxx",129,"",0,""
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: PR[2]: +COLP: "8xxxxxxxxx",129,"",0,""
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: PR[0]: OK
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: cbModemDoDial started
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Dial result 'OK'
kts
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Re: Нет ответа 200(ОК)

Сообщение kts »

2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: UaOnModemAnswer
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: UaOnModemAnswer: answer call with OK.
2017-09-12 15:50:10 User.Notice 192.168.1.100 Dec 1 20:31:06 kts_gate: SIP: 20:31:06.798 pjsua_core.c TX 837 bytes Response msg 200/INVITE/cseq=102 (tdta0x45a178) to UDP 192.168.1.1:5060:<010>SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK580e3c38<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>CSeq: 102 INVITE<013><010>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<013><010>Contact: <sip:0001@192.168.1.100:5061;ob><013><010>Supported: replaces, 100rel, timer, norefersub<013><010>Content-Type: application/sdp<013><010>Content-Length: 290<013><010><013><010>v=0<013><010>o=root 944069454 944069455 IN IP4 192.168.1.100<013><010>s=ktsgate<013><010>c=IN IP4 192.168.1.100<013><010>t=0 0<013><010>a=X-nat:0 Unknown<013><010>m=audio 4000 RTP/AVP 0 101<013><010>a=rtcp:4001 IN IP4 192.168.1.100<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=ptime:20<013><010>a=maxptime:60<013><010>a=sendrecv<013><010><010>--end msg--
2017-09-12 15:50:10 User.Notice 192.168.1.100 Dec 1 20:31:06 kts_gate: SIP: 20:31:06.811 pjsua_core.c RX 413 bytes Request msg ACK/cseq=102 (rdata0x444e3c) from UDP 192.168.1.1:5060:<010>ACK sip:0001@192.168.1.100:5061;ob SIP/2.0<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK1a96d702;rport<013><010>Max-Forwards: 70<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>Contact: <sip:300@192.168.1.1><013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 102 ACK<013><010>User-Agent: Asterisk PBX 1.6.1.17<013><010>Content-Length: 0<013><010><013><010><010>--end msg--
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Call state changed CONNECTING
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: on_call_state finish
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: UaOnModemAnswer finished
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: cbModemDoDial finished
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Call state changed CONFIRMED
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: on_call_state finish
2017-09-12 15:50:10 User.Notice 192.168.1.100 Dec 1 20:31:06 kts_gate: SIP: 20:31:06.856 pjsua_core.c RX 818 bytes Request msg INVITE/cseq=103 (rdata0x444e3c) from UDP 192.168.1.1:5060:<010>INVITE sip:0001@192.168.1.100:5061;ob SIP/2.0<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0746825d;rport<013><010>Max-Forwards: 70<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>Contact: <sip:300@192.168.1.1><013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 103 INVITE<013><010>User-Agent: Asterisk PBX 1.6.1.17<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<013><010>Supported: replaces, timer<013><010>Content-Type: application/sdp<013><010>Content-Length: 264<013><010><013><010>v=0<013><010>o=root 517671258 517671259 IN IP4 192.168.1.105<013><010>s=Asterisk PBX 1.6.1.17<013><010>c=IN IP4 192.168.1.105<013><010>t=0 0<013><010>m=audio 16400 RTP/AVP 0 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=silenceSupp:off - - - -<013><010>a=ptime:20<013><010>a=sendrecv<013><010><010>--end msg--
kts
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Зарегистрирован: 18 янв 2008, 13:36

Re: Нет ответа 200(ОК)

Сообщение kts »

2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Call received updated media offer
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Call remote NAT type is 0 (Unknown)
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Media session for call is destroyed
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Attach media.
2017-09-12 15:50:10 User.Debug 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Set Volume (Rx: 1)(Tx: 1)
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: CODEC: PCMU, payload: 0
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Codec: PCMU ptime: 20. (Vinetic ID: 8).
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Media updates, stream #1 PCMU (sendrecv)
2017-09-12 15:50:10 User.Info 192.168.1.100 Dec 1 20:31:06 kts_gate: 01: Media for Call is active
2017-09-12 15:50:10 User.Notice 192.168.1.100 Dec 1 20:31:06 kts_gate: SIP: 20:31:06.942 pjsua_core.c TX 837 bytes Response msg 200/INVITE/cseq=103 (tdta0x478248) to UDP 192.168.1.1:5060:<010>SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK0746825d<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>CSeq: 103 INVITE<013><010>Contact: <sip:0001@192.168.1.100:5061;ob><013><010>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<013><010>Supported: replaces, 100rel, timer, norefersub<013><010>Content-Type: application/sdp<013><010>Content-Length: 290<013><010><013><010>v=0<013><010>o=root 944069454 944069456 IN IP4 192.168.1.100<013><010>s=ktsgate<013><010>c=IN IP4 192.168.1.100<013><010>t=0 0<013><010>a=X-nat:0 Unknown<013><010>m=audio 4000 RTP/AVP 0 101<013><010>a=rtcp:4001 IN IP4 192.168.1.100<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-16<013><010>a=ptime:20<013><010>a=maxptime:60<013><010>a=sendrecv<013><010><010>--end msg--
2017-09-12 15:50:10 User.Notice 192.168.1.100 Dec 1 20:31:06 kts_gate: SIP: 20:31:06.959 pjsua_core.c RX 413 bytes Request msg ACK/cseq=103 (rdata0x444e3c) from UDP 192.168.1.1:5060:<010>ACK sip:0001@192.168.1.100:5061;ob SIP/2.0<013><010>Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK36e03726;rport<013><010>Max-Forwards: 70<013><010>From: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>To: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>Contact: <sip:300@192.168.1.1><013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 103 ACK<013><010>User-Agent: Asterisk PBX 1.6.1.17<013><010>Content-Length: 0<013><010><013><010><010>--end msg--
2017-09-12 15:50:11 User.Info 192.168.1.100 Dec 1 20:31:08 kts_gate: 01: Run timer "monitor"
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: PR[0]: NO CARRIER
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: OnHangupCall
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Stop timer "request_cid"
2017-09-12 15:50:16 User.Debug 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Hangup call
2017-09-12 15:50:16 User.Notice 192.168.1.100 Dec 1 20:31:13 kts_gate: SIP: 20:31:13.179 pjsua_core.c TX 395 bytes Request msg BYE/cseq=1510 (tdta0x45a178) to UDP 192.168.1.1:5060:<010>BYE sip:300@192.168.1.1 SIP/2.0<013><010>Via: SIP/2.0/UDP 192.168.1.100:5061;rport;branch=z9hG4bKPjGlTyLEokWdfpcRMVwPJPJp5-ZZpMpGUC<013><010>Max-Forwards: 70<013><010>From: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>To: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 1510 BYE<013><010>User-Agent: GSM Gateway v2.2<013><010>Content-Length: 0<013><010><013><010><010>--end msg--
2017-09-12 15:50:16 User.Notice 192.168.1.100 Dec 1 20:31:13 kts_gate: SIP: 20:31:13.185 sip_endpoint.c Processing incoming message: Response msg 200/BYE/cseq=1510 (rdata0x444e3c)
kts
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Сообщения: 183
Зарегистрирован: 18 янв 2008, 13:36

Re: Нет ответа 200(ОК)

Сообщение kts »

2017-09-12 15:50:16 User.Notice 192.168.1.100 Dec 1 20:31:13 kts_gate: SIP: 20:31:13.186 pjsua_core.c RX 490 bytes Response msg 200/BYE/cseq=1510 (rdata0x444e3c) from UDP 192.168.1.1:5060:<010>SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bKPjGlTyLEokWdfpcRMVwPJPJp5-ZZpMpGUC;received=192.168.1.100;rport=5061<013><010>From: <sip:8xxxxxxxxx@192.168.1.100:5061>;tag=-95VvmKKZ5IPsW6lqlZQYegfapGHBl3g<013><010>To: "300" <sip:300@192.168.1.1>;tag=as1b52b342<013><010>Call-ID: 713651ab73a205d93eb828072025487f@192.168.1.1<013><010>CSeq: 1510 BYE<013><010>Server: Asterisk PBX 1.6.1.17<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<013><010>Supported: replaces, timer<013><010>Content-Length: 0<013><010><013><010><010>--end msg--
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: PR[1]: NO CARRIER
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: OnHangupCall
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Stop timer "request_cid"
2017-09-12 15:50:16 User.Debug 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Hangup call
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: on_call_state: Call is inactive, call to ChannelHangup.
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: DoHangup call
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Stop timer "request_cid"
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: AddCommand: ATH (imm: 1 q: 0)
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: WriteMuxCmd(0): ATH
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: AddCommand: AT+CHUP (imm: 1 q: 0)
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: WriteMuxCmd(0): AT+CHUP
2017-09-12 15:50:16 User.Debug 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Call is DISCONNECTED [reason=200 (Normal call clearing)]. Connected duration: 6 sec
2017-09-12 15:50:16 User.Debug 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: on_call_state finish
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Media session for call is destroyed
2017-09-12 15:50:16 User.Warning 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Invalid call in pjsua_call_hangup()
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: PR[2]: NO CARRIER
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: OnHangupCall
2017-09-12 15:50:16 User.Info 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: Stop timer "request_cid"
2017-09-12 15:50:16 User.Warning 192.168.1.100 Dec 1 20:31:13 kts_gate: 01: OnModemHangup: call not found.
2017-09-12 15:50:16 User.Notice 192.168.1.100 Dec 1 20:31:13 kts_gate: SIP: 20:31:13.268 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=102 (rdata0x44d12c)
Avantelecom
Сообщения: 7
Зарегистрирован: 08 сен 2017, 13:50

Re: Нет ответа 200(ОК)

Сообщение Avantelecom »

Тогда я вышлю свои как ситуация возникнет.
Мне ваши логи не о чем не говорят:)
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