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Перезагрузка AP1000 при входящем звонке http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=1257 |
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Автор: | OlegZeml [ 04 мар 2010, 14:36 ] |
Заголовок сообщения: | Перезагрузка AP1000 при входящем звонке |
Привет Всем. Настроил AP1000 подключение по SIP к провайдеру (ЮТК). Исходящий звонок - без проблем. Входящий звонок снаружи - проходит 1 из 10 или 15. В остальных случаях при снятии трубки на addpac тишина, у вызывающего абонента продолжаются гудки вызова. После того как трубка опущена, еще секунд 15 продолжает мигать лампочка порта FXS будто идет разговор, а потом addpac перезагружается. Если подключить ЮТК к Asterisk , то все нормально. Раньше сам аппарат тоже раньше работал с Asterisk. Что может быть? Куда копать? ! version 8.30U ! hostname AP1000 ! ! no bridge spanning-tree ! dhcp-list 1 type server dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128 ! ! no ip-share enable ip-share interface net-side ether0.0 ip-share interface local-side ether1.0 ! interface ether0.0 ip address 10.9.4.11 255.255.255.0 ! interface ether1.0 ip address 192.168.1.123 255.255.255.0 ! snmp name AP1000 snmp enable-trap dn-register 300 forcely-block ! no arp reset ! route 10.10.10.0 255.255.255.0 10.9.4.9 ! ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol t38 redundancy 0 fax rate 9600 h323 call start fast h323 call tunnel enable busyout monitor gatekeeper busyout monitor sip-server no busyout monitor callagent busyout monitor voip-interface ! ! ! Voice port configuration. ! ! FXS voice-port 0/0 caller-id enable ! ! ! FXS voice-port 0/1 caller-id enable ! ! ! FXS voice-port 0/2 caller-id enable ! ! ! FXS voice-port 0/3 caller-id enable ! ! ! ! ! Pots peer configuration. ! dial-peer voice 10 pots destination-pattern 2687023 port 0/0 ! ! ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target sip-server session protocol sip voice-class codec 1 no vad codec-variant g7231 standard dtmf-relay rtp-2833 ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.10.9.4.11 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g729 codec preference 2 g711alaw codec preference 3 g711ulaw codec preference 4 g726r32 ! ! ! ! SIP UA configuration. ! sip-ua sip-username ******** sip-password ********* sip-server 10.10.10.100 register e164 ! ! ! MGCP configuration. ! mgcp no codec vad ! ! ! Tones ! Дебаг успешного звонка. AP1000# Received SIP PDU from ( 10.10.10.100:5070 ) INVITE sip:2687069@10.9.4.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3 Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff To: <sip:2687069@10.9.4.11;user=phone> CSeq: 1 INVITE Contact: <sip:Anonymous@10.10.10.100:5070> Supported: 100rel User-Agent: Huawei SoftX3000 V300R006B06D061 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER Content-Length: 248 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 3293963 3293963 IN IP4 10.10.10.100 s=Sip Call c=IN IP4 10.10.10.10 t=0 0 m=audio 29908 RTP/AVP 18 8 97 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=no Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff To: <sip:2687069@10.9.4.11;user=phone> Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 1 <Call 3> : ****************** Call Created status(InitiatedByNet) ******************* 2 <SIP 3> : Receive INVITE Request 3 <NetCon 3> : Found inbound voip peer by dest-pattern id(1000) 4 <Call 3> : From Net - calledParty(2687069) callingParty(Anonymous) 5 <Call 3> : MatchedPerfect 6 <Call 3> : MatchAllProcess After Sorted <0> id(10) dest(2687069) prefer(0) selected(0) 7 <Call 3> : Initiate callee with dial-peer(2687069) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 8 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS) [1451.575] RTA(0/0/0) Rx CC_RING_REQ [80 18 01 08 30 33 30 34 31 37 30 34 04 01 4f 07 09 41 6e 6f 6e 79 6d 6f 75 73 ] peerId(-1) [1451.575] VM(0/0/0) DaTime [L=8] 30 33 30 34 31 37 30 34 [1451.575] VM(0/0/0) CgNoNu [L=1] 4f [1451.575] VM(0/0/0) CgName [L=9] 41 6e 6f 6e 79 6d 6f 75 73 [1451.575] VM(0/0/0) Line Reverse [1451.575] VM(0/0/0) Start ring actv 9 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(3) [1451.580] VM(0/0/0) set T38 mode STD [1451.580] VM(0/0/0) Fax rate 9600 10 <SIP 3> : SetAlerting Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4 Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Contact: sip:2687069@10.9.4.11 Content-Length: 0 [1452.575] VM(0/0/0) Gen ring idle [1453.075] VM(0/0/0) Tx CID enable [1453.075] VM(0/0/0) vopp enable [1453.075] VM(0/0/0) play mute [1453.135] VM(0/0/0) Tx CID ON [1453.180] VM(0/0/0) Rx CID_ACK [1453.180] VM(0/0/0) Tx CID DATA [L=54] 80 01 18 02 01 05 08 06 30 08 33 08 30 08 34 08 31 07 37 07 30 07 34 07 04 05 01 06 4f 0c 07 05 09 06 41 0b 6e 0b 6f 0b 6e 0b 79 0b 6d 0b 6f 0b 75 0b 73 0b 00 0f [1454.180] VM(0/0/0) Tx CID fin [1454.180] VM(0/0/0) vopp idle [1454.180] VM(0/0/0) VoPP ready [1454.365] VM(0/0/0) vmOffHook [1454.425] VM(0/0/0) vmTmoOffHook [1454.425] VM(0/0/0) Line Forward [1454.485] VM(0/0/0) vmTmoOffHook [1454.485] VM(0/0/0) Rx OffHook [1454.485] VM(0/0/0) vopp enable [1454.485] VM(0/0/0) Fax enable [1454.485] VM(0/0/0) play mute [1454.485] VM(0/0/0) Tx CONNECT_CNF 11 <Call 3> : Connected from(0) [1454.485] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [1454.485] VM(0/0/0) VAD disable [1454.485] VM(0/0/0) SID enable by CCC 12 <SIP 3> : SetConnected 13 <SIP 3> : Add Local Audio MediaFormat : 18 [1454.490] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [1454.490] VM(0/0/0) VAD disable Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4 Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:2687069@10.9.4.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 250 v=0 o=2687069 1267722296 1267722296 IN IP4 10.9.4.11 s=AddPac Gateway SDP c=IN IP4 10.9.4.11 t=1267722296 0 m=audio 23002 RTP/AVP 18 97 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000/1 a=fmtp:97 0-15 a=ptime:20 [1454.525] RTA(0/0/0) Rx RS_LISTEN_REQ callId=3 ssId=1 G729A peer=10.10.10.10 mp=23002/23003 hp=29908/29909 [1454.525] VM(0/0/0) codec same G729A [1454.525] RTA(0/0/0) Rx RS_OPEN_REQ callId=3 ssId=1 G729A peer=10.10.10.10 mp=23002/23003 hp=29908/29909 [1454.525] VM(0/0/0) codec same G729A [1454.525] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [1454.525] VM(0/0/0) DTMF_RTP_RFC2833 enable [1454.525] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x61, RxPT=0x61 Received SIP PDU from ( 10.10.10.100:5070 ) ACK sip:2687069@10.9.4.11 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bKfbf01e6fc Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 14 <SIP 3> : ACK received 15 <SIP 3> : Receive ACK Request 16 <SIP 3> : Set Terminated Success for 1 INVITE [1457.760] RTA(0/0/0) RTP play loss, xCnt=1 [1458.020] RTA(0/0/0) RTP play loss, xCnt=1 [1458.030] RTA(0/0/0) RTP play loss, xCnt=1 [1458.780] RTA(0/0/0) RTP play loss, xCnt=1 [1458.790] RTA(0/0/0) RTP play loss, xCnt=1 [1459.960] RTA(0/0/0) RTP play loss, xCnt=1 [1460.740] RTA(0/0/0) RTP play loss, xCnt=1 [1460.750] RTA(0/0/0) RTP play loss, xCnt=1 [1463.420] RTA(0/0/0) RTP play loss, xCnt=1 [1464.580] RTA(0/0/0) RTP play loss, xCnt=1 [1466.120] RTA(0/0/0) RTP play loss, xCnt=1 [1466.660] RTA(0/0/0) RTP play loss, xCnt=1 [1466.670] RTA(0/0/0) RTP play loss, xCnt=1 [1468.240] RTA(0/0/0) RTP play loss, xCnt=1 [1468.380] RTA(0/0/0) RTP play loss, xCnt=1 [1469.820] RTA(0/0/0) RTP play loss, xCnt=1 [1469.830] RTA(0/0/0) RTP play loss, xCnt=1 [1470.875] VM(0/0/0) vmOnHook [1470.925] VM(0/0/0) vmTmoOnHook [1470.975] VM(0/0/0) vmTmoOnHook [1471.020] RTA(0/0/0) RTP play loss, xCnt=1 [1471.025] VM(0/0/0) vmTmoOnHook [1471.030] RTA(0/0/0) RTP play loss, xCnt=1 [1471.075] VM(0/0/0) vmTmoOnHook [1471.125] VM(0/0/0) vmTmoOnHook [1471.175] VM(0/0/0) vmTmoOnHook [1471.225] VM(0/0/0) vmTmoOnHook [1471.275] VM(0/0/0) vmTmoOnHook [1471.300] RTA(0/0/0) RTP play loss, xCnt=1 [1471.310] RTA(0/0/0) RTP play loss, xCnt=1 [1471.325] VM(0/0/0) vmTmoOnHook [1471.375] VM(0/0/0) vmTmoOnHook [1471.425] VM(0/0/0) vmTmoOnHook [1471.460] RTA(0/0/0) RTP play loss, xCnt=1 [1471.470] RTA(0/0/0) RTP play loss, xCnt=1 [1471.475] VM(0/0/0) vmTmoOnHook [1471.525] VM(0/0/0) vmTmoOnHook [1471.575] VM(0/0/0) vmTmoOnHook [1471.575] VM(0/0/0) Rx OnHook [1471.575] VM(0/0/0) vopp idle [1471.575] VM(0/0/0) VoPP ready [1471.575] VM(0/0/0) Tx DISCONN_CNF 17 <CEP 000000> : Disconnected(16) at Busy 18 <Call 3> : Terminated from(0) this(Local:CallClear) before(NULL) forced(0) 19 <SIP 3> : ReleaseWithBYE 20 <SIP 3> : Send BYE Request Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 BYE sip:Anonymous@10.10.10.100:5070 SIP/2.0 Via: SIP/2.0/UDP 10.9.4.11:5060;branch=z9hG4bK354be505a41 From: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4 To: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 CSeq: 1 BYE Date: Thu, 04 Mar 2010 17:05:13 GMT User-Agent: AddPac SIP Gateway Contact: <sip:2687069@10.9.4.11> Content-Length: 0 Max-Forwards: 70 [1471.595] RTA(0/0/0) Rx RS_CLOSE_REQ callId=3 ssId=1 dir=reve [1471.595] RTA(0/0/0) Rx RS_CLOSE_REQ callId=3 ssId=1 dir=forw [1471.595] RTA(0/0/0) close Media socket [1471.595] RTA(0/0/0) close RTCP socket 21 <NetEP 3> : Call TO <Anonymous> terminated reason(Local:CallClear) 22 <CEP 000000> : DisconnectCall at Idle Received SIP PDU from ( 10.10.10.100:5070 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.9.4.11:5060;branch=z9hG4bK354be505a41 Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100 From: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4 To: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff CSeq: 1 BYE Content-Length: 0 23 <SIP 3> : Receive 200 OK 24 <SIP 3> : Transaction (1 BYE) completed 25 <SIP 3> : Set Terminated Success for 1 BYE Дебаг до перезагрузки: Received SIP PDU from ( 10.10.10.100:5070 ) INVITE sip:2687069@10.9.4.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60 Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c To: <sip:2687069@10.9.4.11;user=phone> CSeq: 1 INVITE Contact: <sip:Anonymous@10.10.10.100:5070> Supported: 100rel User-Agent: Huawei SoftX3000 V300R006B06D061 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER Content-Length: 248 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 3293976 3293976 IN IP4 10.10.10.100 s=Sip Call c=IN IP4 10.10.10.10 t=0 0 m=audio 29950 RTP/AVP 18 8 97 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=no Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c To: <sip:2687069@10.9.4.11;user=phone> Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100 CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 26 <Call 4> : ****************** Call Created status(InitiatedByNet) ******************* 27 <SIP 4> : Receive INVITE Request 28 <NetCon 4> : Found inbound voip peer by dest-pattern id(1000) 29 <Call 4> : From Net - calledParty(2687069) callingParty(Anonymous) 30 <Call 4> : MatchedPerfect 31 <Call 4> : MatchAllProcess After Sorted <0> id(10) dest(2687069) prefer(0) selected(1) 32 <Call 4> : Initiate callee with dial-peer(2687069) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 33 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS) [1488.775] RTA(0/0/0) Rx CC_RING_REQ [80 18 01 08 30 33 30 34 31 37 30 35 04 01 4f 07 09 41 6e 6f 6e 79 6d 6f 75 73 ] peerId(-1) [1488.775] VM(0/0/0) DaTime [L=8] 30 33 30 34 31 37 30 35 [1488.775] VM(0/0/0) CgNoNu [L=1] 4f [1488.775] VM(0/0/0) CgName [L=9] 41 6e 6f 6e 79 6d 6f 75 73 [1488.775] VM(0/0/0) Line Reverse [1488.775] VM(0/0/0) Start ring actv 34 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(4) [1488.780] VM(0/0/0) set T38 mode STD [1488.780] VM(0/0/0) Fax rate 9600 35 <SIP 4> : SetAlerting Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60 From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c To: <sip:2687069@10.9.4.11;user=phone>;tag=5a4b2706a4 Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100 CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Contact: sip:2687069@10.9.4.11 Content-Length: 0 |
Автор: | Geniu$$ [ 05 мар 2010, 08:53 ] |
Заголовок сообщения: | |
Попробуйте перепрошить С питанием все нормально? |
Автор: | OlegZeml [ 05 мар 2010, 09:12 ] |
Заголовок сообщения: | |
С питание все нормально Прошивка 8_30U |
Автор: | OlegZeml [ 23 мар 2010, 09:02 ] |
Заголовок сообщения: | |
В общем проблема решилась заменой блока питания.... |
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