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AddPac IP100 и факсы http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=3403 |
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Автор: | vint [ 10 апр 2013, 13:33 ] |
Заголовок сообщения: | AddPac IP100 и факсы |
Добрый день! К AP-IP100 в порт fxs подключен факс. AP зарегистрирован на IP-АТС по SIP. Так вот факсы не ходят в обе стороны! Как проверялось: С факса №305, подключенный к AP-IP100 набирал номер факса (№111), подключенного к внутреннему порту атс. Вызов приходит, голос проключается, но при старте факса абонентом №305, абонент №111 даже не слышит сигнала факса. При звонке в обратном направлении абонент №305 слышит сигнал факса абонента №111, но факсы все равно не проходят! Используется кодек G.711. вот конфиг AP-IP100: Current configuration: ! version 8.41.083 ! hostname IP100 ! IP PHONE OSD configuration. ! osd language english network signaling sip network sscp disable phone lcd-type graphic phone ring-type 1 phone volume ring 4 phone volume input 5 phone volume output 5 phone volume micbooster disable phone auto-hook-on disable phone display-name 302 phone voice-codec 0 phone dnd-mode silence phone pbx-mode general phone auto-answer disable phone save-mode always phone forward-status disable phone conference-status disable phone password 2337 phone password-status disable phone admin-lock factory status disable phone admin-lock internet status disable phone admin-lock voip status enable phone admin-lock service status disable phone admin-lock auto-upgrade status disable phone admin-lock sscp status disable phone privacy-password 0000 phone privacy-status disable phone privacy-lock menu status disable phone privacy-lock incoming status disable phone privacy-lock outgoing status disable ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol bypass fax rate 9600 h323 call start fast h323 call tunnel enable timeout tmohdt 300 call-barring unconfigured-ip-address ! ! ! Voice port configuration. ! ! SPEECH voice-port 0/0 ! ! ! FXS voice-port 0/1 caller-id enable ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 302 port 0/0 ! dial-peer voice 1 pots destination-pattern 305 port 0/1 ! ! Voip peer configuration. ! dial-peer voice 100 voip destination-pattern 1.. session target ras no vad dtmf-relay h245-alphanumeric ! dial-peer voice 200 voip destination-pattern 2.. session target ras no vad dtmf-relay h245-alphanumeric ! dial-peer voice 1001 voip destination-pattern T session target sip-server session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 huntstop ! dial-peer voice 1002 voip destination-pattern T session target ras voice-class codec 0 no vad dtmf-relay rtp-2833 preference 1 huntstop ! ! ! dial-peer call-hold h dial-peer call-transfer h ! ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.200.173 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711alaw codec preference 2 g711alaw codec preference 3 g711alaw codec preference 4 g711alaw ! voice class codec 1 ! ! ! ! SIP UA configuration. ! sip-ua fault-tolerance 10 500 sip-username 305 sip-password ********* sip-server 192.168.ххх.ххх rport enable media-channel early register e164 ! ! ! Tones ! ! ! ! line console ! line vty ! ! sms quota 30 ! end А ВОТ ДЕБАГ ЗВОНКА: IP100# IP100# [17059.560] VM(0/1/0) vmOffHook [17059.620] VM(0/1/0) vmTmoOffHook [17059.620] VM(0/1/0) Rx OffHook [17059.620] VP(0/1/0) use line [17059.620] VP(0/0/0) add line [17059.620] VP(0/1/0) enable Fax, disable Modem [17059.620] VP(0/1/0) open channel [17059.620] VM(0/1/0) Tx OFFHOOK_IND [17059.620] VM(0/1/0) play Dial tone [17059.620] VP(0/1/0) Tx IBS signal 6/0 [17059.620] VP(0/1/0) Tx IBS dir 0 80 <CEP 000100> : Call Received 81 <CEP 000100> : Call Initiated : calledNumber() crv(0) total(0) 82 <Call 62> : ****** Call Created status(InitiatedByFXS) ver(8.28:2006-02-06-00-00) time(17014) **** 83 <CEP 000100> : Calling number(305) 84 <CEP 000100> : Call id(76420000-55fa-0017-8089-0002a4082742) callNum(62) [17066.080] VM(0/1/0) Tx DIGIT_IND '1' [17066.080] VM(0/1/0) play mute [17066.080] VP(0/1/0) Tx IBS signal 2/0 [17066.080] VP(0/1/0) Tx IBS dir 0 85 <Call 62> : Digit(1) at InitiatedByFXS 86 <Call 62> : MatchedAll [17066.150] VP(0/1/0) GeneralEvent IBS gen end [17066.430] VM(0/1/0) Tx DIGIT_IND '1' 87 <Call 62> : Digit(1) at CalleeDeterminedWaitDigit 88 <Call 62> : MatchedAll [17066.780] VM(0/1/0) Tx DIGIT_IND '1' 89 <Call 62> : Digit(1) at CalleeDeterminedWaitDigit 90 <Call 62> : MatchedPerfect 91 <Call 62> : MatchAllProcess After Sorted <0> id(100) dest(1..) prefer(0) selected(34) <1> id(1001) dest(T) prefer(0) selected(0) <2> id(1002) dest(T) prefer(1) selected(0) 92 <Call 62> : Initiate callee with dial-peer(1..) status(CalleeDeterminedAll) id(76420000-55fa-0017-8089-0002a4082742) 93 <NetEP 62> : InitiateOutCall: calledNum(111) callingNum(305) target(ras) 94 <Call 62> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(76420000-55fa-0017-8089-0002a4082742) 95 <NetEP 62> : InitiateOutCall: calledNum(111) callingNum(305) target(sip-server) 96 <NetEP 62> : DoCall: calledAddr(sip:111@192.168.200.230:5060) callingAddr(305) [17066.780] VM(0/1/0) set T38 disable [17066.780] VM(0/1/0) Fax rate 9600 97 <SIP 0> : No authentication information available 98 <SIP 62> : Send INVITE Request Sending SIP PDU to ( 192.168.200.230:5060 ) from 5060 INVITE sip:111@192.168.200.230 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa452 From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230> Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 52 INVITE Supported: replaces, timer, 100rel, early-session Min-SE: 1800 Date: Thu, 01 Jan 2009 04:43:42 GMT Session-Expires: 1800 User-Agent: AddPac SIP Gateway Contact: <sip:305@192.168.200.173> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO Allow-Events: talk, hold, conference Content-Type: application/sdp Content-Length: 217 Max-Forwards: 70 v=0 o=305 17022 17022 IN IP4 192.168.200.173 s=AddPac Gateway SDP c=IN IP4 192.168.200.173 t=0 0 m=audio 23162 RTP/AVP 8 101 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [17066.790] RTA(0/1/0) Rx RS_LISTEN_REQ callId=62 ssId=1 G711U peer=0.0.0.0 mp=23162/23163 hp=0/0 [17066.790] VM(0/1/0) codec same G711U Received SIP PDU from ( 192.168.200.230:5060 ) SIP/2.0 100 Trying From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230> Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 52 INVITE Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa452 Contact: <sip:111@192.168.200.230:5060;transport=udp> Content-Length: 0 99 <SIP 62> : Receive 100 Trying 100 <SIP 62> : Transaction (52 INVITE) proceeding Received SIP PDU from ( 192.168.200.230:5060 ) SIP/2.0 180 Ringing From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230>;tag=41a42318-e6c8a8c0-13c4-50017-5165c846-58a94b64-5165c846 Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 52 INVITE P-Asserted-Identity: <sip:111@192.168.200.230> Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa452 Contact: <sip:111@192.168.200.230:5060;transport=udp> Content-Length: 0 101 <SIP 62> : Receive 180 Ringing 102 <SIP 62> : Transaction (52 INVITE) proceeding 103 <SIP 62> : Received Session Progress response 104 <SIP 62> : No session information [17066.915] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [17066.915] VM(0/1/0) DTMF enable [17066.915] VM(0/1/0) DTMF_RTP_RFC2833 enable [17066.915] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65 105 <Call 62> : Alert from(fffffffe) pseudo(0) inband(0) status(CalleeInitiated) [17066.915] RTA(0/1/0) Rx CC_ALERT_RSP peerId(0/0/0) [17066.915] VM(0/1/0) play RingBack tone [17066.915] VP(0/1/0) Tx IBS signal 6/1 [17066.915] VP(0/1/0) Tx IBS dir 0 Received SIP PDU from ( 192.168.200.230:5060 ) SIP/2.0 200 OK From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230>;tag=41a42318-e6c8a8c0-13c4-50017-5165c846-58a94b64-5165c846 Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 52 INVITE P-Asserted-Identity: <sip:111@192.168.200.230> Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa452 Contact: <sip:111@192.168.200.230:5060;transport=udp> Supported: 100rel Content-Type: application/sdp Content-Length: 224 v=0 o=SAMSUNG_SIP_GATEWAY 4120272458 0 IN IP4 192.168.200.230 s=SIP_CALL c=IN IP4 192.168.200.230 t=0 0 m=audio 30010 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 106 <SIP 62> : Receive 200 OK 107 <SIP 62> : Received INVITE OK response 108 <SIP 62> : Send ACK Request Sending SIP PDU to ( 192.168.200.230:5060 ) from 5060 ACK sip:111@192.168.200.230;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa452 From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230>;tag=41a42318-e6c8a8c0-13c4-50017-5165c846-58a94b64-5165c846 Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 52 ACK Content-Length: 0 Max-Forwards: 70 109 <SIP 62> : Get SIP Audio MediaFormat : 8 [17069.865] RTA(0/1/0) Rx RS_OPEN_REQ callId=62 ssId=1 G711A peer=192.168.200.230 mp=23162/23163 hp=30010/30011 [17069.865] VM(0/1/0) vopp idle [17069.865] VP(0/1/0) close channel [17069.865] VM(0/1/0) start codec replace timer to G711A 110 <Call 62> : Connected from(fffffffe) [17069.870] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [17069.870] VM(0/1/0) VAD disable [17069.870] VP(0/1/0) ignore notEnabledCh updating VAD 0 [17069.870] VM(0/1/0) SID enable by CCC [17069.870] RTA(0/1/0) Rx CC_CONNECT_RSP peerId(0/0/0) [17069.870] VM(0/1/0) T38 Fax disabled 111 <NetEP 62> : Call with sip:111@192.168.200.230 established [17069.870] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [17069.870] VM(0/1/0) DTMF enable [17069.870] VM(0/1/0) DTMF_RTP_RFC2833 enable [17069.870] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65 112 <SIP 62> : Check Event Relation 113 <SIP 62> : Set Terminated Success for 52 INVITE [17069.925] VP(0/1/0) enable Fax, disable Modem [17069.925] VP(0/1/0) open channel [17069.925] VM(0/1/0) codec replaced to G711A [17069.925] VM(0/1/0) play mute [17069.925] VP(0/1/0) Tx IBS signal 2/0 [17069.925] VP(0/1/0) Tx IBS dir 0 [17070.000] VP(0/1/0) GeneralEvent IBS gen end [17070.095] VM(0/1/0) codec same G711A [17070.095] VM(0/1/0) Rx RTP replace codec to G711A [17077.880] VP(0/1/0) detect FaxModemEvent FB=0 in=2 out=0 [17077.880] VP(0/1/0) no ActFaxRelay by T38 disable [17077.880] VP(0/1/0) GeneralEvent index=9 : state=FAX Reason=21 [17077.885] VP(0/1/0) FaxRelayStatus len 22 [17080.810] VP(0/1/0) detect FaxModemEvent FB=0 in=6 out=0 [17080.850] VP(0/1/0) FaxRelayStatus len 22 [17084.160] VP(0/1/0) FaxRelayStatus len 22 [17088.750] VP(0/1/0) FaxRelayStatus len 22 [17092.100] VP(0/1/0) FaxRelayStatus len 22 [17096.690] VP(0/1/0) FaxRelayStatus len 22 114 <Time 0> : SIP Dialog Expire timer timeout. [17100.035] VP(0/1/0) FaxRelayStatus len 22 [17113.830] VP(0/1/0) FaxRelayStatus len 22 [17115.050] VP(0/1/0) FaxRelayStatus len 22 [17115.535] VM(0/1/0) vmOnHook [17115.585] VM(0/1/0) vmTmoOnHook [17115.635] VM(0/1/0) vmTmoOnHook [17115.685] VM(0/1/0) vmTmoOnHook [17115.735] VM(0/1/0) vmTmoOnHook [17115.785] VM(0/1/0) vmTmoOnHook [17115.835] VM(0/1/0) vmTmoOnHook [17115.885] VM(0/1/0) vmTmoOnHook [17115.935] VM(0/1/0) vmTmoOnHook [17115.985] VM(0/1/0) vmTmoOnHook [17116.035] VM(0/1/0) vmTmoOnHook [17116.085] VM(0/1/0) vmTmoOnHook [17116.135] VM(0/1/0) vmTmoOnHook [17116.185] VM(0/1/0) vmTmoOnHook [17116.235] VM(0/1/0) vmTmoOnHook [17116.235] VM(0/1/0) Rx OnHook [17116.235] VM(0/1/0) vopp idle [17116.235] VP(0/1/0) close channel [17116.235] VM(0/1/0) Tx DISCONN_CNF 115 <CEP 000100> : Disconnected(16) at Busy 116 <Call 62> : Terminated from(100) this(Local:CallClear) before(NULL) forced(0) time(17071) 117 <CEP 000100> : DisconnectCall at Idle 118 <SIP 62> : ReleaseWithBYE 119 <SIP 62> : Send BYE Request Sending SIP PDU to ( 192.168.200.230:5060 ) from 5060 BYE sip:111@192.168.200.230;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa453 From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230>;tag=41a42318-e6c8a8c0-13c4-50017-5165c846-58a94b64-5165c846 Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 53 BYE Date: Thu, 01 Jan 2009 04:44:31 GMT User-Agent: AddPac SIP Gateway Contact: <sip:305@192.168.200.173> Content-Length: 0 Max-Forwards: 70 [17116.240] RTA(0/1/0) Rx RS_CLOSE_REQ callId=62 ssId=1 dir=all [17116.240] RTA(0/1/0) close Media socket [17116.240] RTA(0/1/0) close RTCP socket 120 <NetEP 62> : Call TO <sip:111@192.168.200.230> terminated reason(Local:CallClear) Received SIP PDU from ( 192.168.200.230:5060 ) SIP/2.0 200 OK From: <sip:305@192.168.200.230>;tag=7e004f8aa4 To: <sip:111@192.168.200.230>;tag=41a42318-e6c8a8c0-13c4-50017-5165c846-58a94b64-5165c846 Call-ID: 7e420000-0f9b-4ff6-808a-0002a4082742@192.168.200.173 CSeq: 53 BYE Via: SIP/2.0/UDP 192.168.200.173:5060;branch=z9hG4bK7e004f8aa453 Content-Length: 0 Бьюсь уже два дня а где затык понять не могу! Заранее спасибо! |
Автор: | genal [ 12 апр 2013, 09:20 ] |
Заголовок сообщения: | Re: AddPac IP100 и факсы |
Менять следующие параметры пробовали?: fax protocol bypass fax rate 9600 |
Автор: | Vlad1965 [ 24 апр 2013, 07:52 ] |
Заголовок сообщения: | Re: AddPac IP100 и факсы |
1. >С факса №305, подключенный к AP-IP100 набирал номер факса (№111), >подключенного к внутреннему порту атс. Вызов приходит, голос >проключается, но при старте факса абонентом №305, абонент №111 даже >не слышит сигнала факса. Похоже что где-то включается работа по протоколу T38 - попробуйте на IP-АТС отключить службу (если включена) работы с факсами , в настройки аддпака: voice service voip fax protocol bypass 2. >При звонке в обратном направлении абонент №305 слышит сигнал факса >абонента №111, но факсы все равно не проходят! Попробуйте на аддпаке: voice-port 0/1 no comfort-noise no echo-cancel P.S. ну и естесствено кодек G711 - здесь все Ок |
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