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Факсы ходят только в одну сторону http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2498 |
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Автор: | Silver-D [ 27 сен 2011, 15:24 ] |
Заголовок сообщения: | Факсы ходят только в одну сторону |
Всем добрый день. Столкнулся с проблемой. Имеем AP200B и SIP. Проблема в том, что факсы ходят только в одну сторону. Т.е. я могу послать факс со шлюза на городской телефон (FAX=>шлюз=>SIP=>городской тел=>FAX). А обратно (FAX=>городской тел=>SIP=>шлюз=>FAX) факсы не идут. Происходит это так: - набираем номер на факсе городского тел - факс (1). - начинает звонить факс на шлюзе - факс (2) - факс (2) отвечает на звонок и начинает пищать для соединения - факс (1) и факс (2) пишет, что -=обнаружен ответ факса=- - и на этом этапе шлюз, вешает линию. Просто берет и отрубает. Что пробовал делать: - менял факсы местами (1) и (2). Без изменений. Т.е. (FAX=>городской тел=>SIP=>шлюз=>FAX) факсы не идут. Шлюз вешает трубку. - всячески менял настройки шлюза. Ничего не помогает. Провайдер Т38 не поддерживает. Факсы только по g711. Конфиг: Цитата: version 8.30V ! hostname AP200 ! ! no bridge spanning-tree ! dhcp-list 0 type server dhcp-list 0 address server interface ether0.0 dhcp-list 0 option dhcp-lease-time 7200 ! dhcp-list 1 type server dhcp-list 1 address server 10.1.1.2 10.1.1.254 255.255.255.0 dhcp-list 1 option dhcp-lease-time 600 ! ! ip-share enable ip-share interface net-side ether0.0 ip-share interface local-side ether1.0 ! interface ether0.0 ip address 192.168.1.20 255.255.255.0 ! interface ether1.0 no ip address ! snmp name AP200B ! no arp reset ! route 0.0.0.0 0.0.0.0 192.168.1.1 ! dnshost nameserver 192.168.1.1 ! pnp-sktelink debug on ! auto-script autorun.inf ! ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol bypass fax rate 9600 h323 call start fast h323 call tunnel enable announcement language english busyout monitor gatekeeper busyout monitor sip-server no busyout monitor callagent busyout monitor voip-interface ! ! ! Voice port configuration. ! ! FXS voice-port 0/0 low-dtmf-gain 0 high-dtmf-gain 0 caller-id enable caller-id type etsi-dtmf ! ! ! FXS voice-port 0/1 caller-id enable ! ! ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 14924 port 0/0 ! ! ! ! Voip peer configuration. ! dial-peer voice 2000 voip destination-pattern T session target sip-server session protocol sip voice-class codec 1 no vad dtmf-relay rtp-2833 ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.1.20 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711alaw codec preference 2 g711ulaw ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw ! voice class codec 1000 codec preference 1 g711alaw codec preference 2 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-username 14924 sip-password xxxxxxxxx sip-server 188.134.000.243 register e164 ! ! ! MGCP configuration. ! mgcp codec g711alaw vad ! ! ! Tones ! ! ! ! Debug: Цитата: 17<Call1>: ****************** Call Created status(InitiatedByNet) *******************
18<SIP1>: Receive INVITE Request 19<NetCon1>: Found inbound voip peer by dest-pattern id(2000) 20<Call1>: From Net - calledParty(14924) callingParty(+781274xxxxx) 21<Call1>: MatchedPerfect 22<Call1>: MatchAllProcess After Sorted <0> id(0) dest(14924) prefer(0) selected(0) 23<Call1>: Initiate callee with dial-peer(14924) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 24<CEP000000>: InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS) [814.800] RTA(0/0/0) Rx CC_RING_REQ [80 ] peerId(-1) [814.800] VM(0/0/0) Rx RingReq CID msg ERR [814.800] VM(0/0/0) Line Reverse [814.800] VM(0/0/0) Start ring actv 25<CEP000000>: Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(1) [814.800] VM(0/0/0) Fax rate 9600 26<SIP1>: SetAlerting Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4204e690;rport From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5 To: <sip:14924@192.168.1.20>;tag=004ee201a4 Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Contact: sip:14924@192.168.1.20 Content-Length: 0 [815.800] VM(0/0/0) Gen ring idle [817.800] VM(0/0/0) Gen ring actv [818.800] VM(0/0/0) Gen ring idle [820.800] VM(0/0/0) Gen ring actv [821.800] VM(0/0/0) Gen ring idle [823.800] VM(0/0/0) Gen ring actv [824.800] VM(0/0/0) Gen ring idle [826.120] VM(0/0/0) vmOffHook [826.180] VM(0/0/0) vmTmoOffHook [826.180] VM(0/0/0) Line Forward [826.205] VM(0/0/0) vmOnHook [826.210] VM(0/0/0) vmOffHook [826.270] VM(0/0/0) vmTmoOffHook [826.270] VM(0/0/0) Rx OffHook [826.270] VM(0/0/0) Modem attribute disable [826.270] VM(0/0/0) Modem attribute G711A [826.270] VM(0/0/0) vopp enable [826.270] VM(0/0/0) Tx CONNECT_CNF 27<Call1>: Connected from(0) [826.270] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [826.270] VM(0/0/0) VAD disable [826.270] VM(0/0/0) SID enable by CCC 28<SIP1>: SetConnected 29<SIP1>: Add Local Audio MediaFormat : 8 [826.280] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [826.280] VM(0/0/0) VAD disable Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4204e690;rport From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5 To: <sip:14924@192.168.1.20>;tag=004ee201a4 Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru CSeq: 102 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:14924@192.168.1.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 236 v=0 o=14924 1317135372 1317135372 IN IP4 192.168.1.20 s=AddPac Gateway SDP c=IN IP4 192.168.1.20 t=1317135372 0 m=audio 23002 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 [826.320] RTA(0/0/0) Rx RS_LISTEN_REQ callId=1 ssId=1 G711A peer=188.134.0.243 mp=23002/23003 hp=13480/13481 [826.320] VM(0/0/0) vopp idle [826.320] VM(0/0/0) start codec replace timer to G711A [826.320] RTA(0/0/0) Rx RS_OPEN_REQ callId=1 ssId=1 G711A peer=188.134.0.243 mp=23002/23003 hp=13480/13481 [826.320] VM(0/0/0) under codec replace to G711A [826.320] VM(0/0/0) V152_modem TxPT=0x64, RxPT=0x64 [826.320] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [826.320] VM(0/0/0) DTMF_RTP_RFC2833 enable [826.320] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65 Received SIP PDU from ( 188.134.0.243:5060 ) ACK sip:14924@192.168.1.20 SIP/2.0 Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK5e989f40;rport From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5 To: <sip:14924@192.168.1.20>;tag=004ee201a4 Contact: <sip:+781274xxxxx@188.134.0.243> Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru CSeq: 102 ACK User-Agent: InterZet PBX Max-Forwards: 70 Content-Length: 0 30<SIP1>: ACK received 31<SIP1>: Receive ACK Request 32<SIP1>: Set Terminated Success for 102 INVITE [826.380] VM(0/0/0) Modem attribute disable [826.380] VM(0/0/0) Modem attribute G711A [826.380] VM(0/0/0) vopp enable [826.380] VM(0/0/0) codec replaced to G711A [826.380] VM(0/0/0) play mute Received SIP PDU from ( 188.134.0.243:5060 ) BYE sip:14924@192.168.1.20 SIP/2.0 Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4a7ba1f6;rport From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5 To: <sip:14924@192.168.1.20>;tag=004ee201a4 Contact: <sip:+781274xxxxx@188.134.0.243> Call-ID: 5af2be469bb330d5de76daa5dbce3f0@voip.interzet.ru CSeq: 103 BYE User-Agent: InterZet PBX Max-Forwards: 70 Content-Length: 0 33<SIP1>: Receive BYE Request Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4a7ba1f6;rport From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5 To: <sip:14924@192.168.1.20>;tag=004ee201a4 Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru CSeq: 103 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 34<SIP1>: ReleaseWithNothing [834.790] RTA(0/0/0) Rx RS_CLOSE_REQ callId=1 ssId=1 dir=reve [834.790] RTA(0/0/0) Rx RS_CLOSE_REQ callId=1 ssId=1 dir=forw [834.790] RTA(0/0/0) close Media socket [834.790] RTA(0/0/0) close RTCP socket 35<Call1>: Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0) 36<CEP000000>: DisconnectCall at Busy 37<CEP000000>: StopSignal [834.795] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_STOP [834.795] VM(0/0/0) play mute 38<CEP000000>: Disconnect (0) [834.795] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0) [834.795] VM(0/0/0) play Reorder tone 39<NetEP1>: Call TO <+781274xxxxx> terminated reason(Remote:CallClear) 40<Time0>: SIP_TREGISTER timer timeout. 41<SIP0>: Adding authentication information 42<SIP35>: Send REGISTER Request Sending SIP PDU to ( 188.134.000.243:5060 ) from 5060 REGISTER sip:188.134.000.243 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 35 REGISTER Date: Tue, 27 Sep 2011 14:56:28 GMT User-Agent: AddPac SIP Gateway Authorization: Digest username="14924", realm="voip.interzet.ru", nonce="75f7a428", uri="sip:188.134.000.243", response="cf373d878a7140e5ddaedfea85b859f5", algorithm=MD5 Contact: <sip:14924@192.168.1.20>;expires=60 Expires: 60 Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 188.134.0.243:5060 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435;received=188.134.30.161 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 35 REGISTER User-Agent: InterZet PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:14924@188.134.0.243> Content-Length: 0 43<SIP35>: Receive 100 Trying 44<SIP35>: Transaction (35 REGISTER) proceeding Received SIP PDU from ( 188.134.0.243:5060 ) SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435;received=188.134.30.161 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243;tag=as217b6d45 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 35 REGISTER User-Agent: InterZet PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:14924@188.134.0.243> WWW-Authenticate: Digest realm="voip.interzet.ru", nonce="661f5f4c" Content-Length: 0 45<SIP35>: Receive 401 Unauthorized 46<SIP35>: Transaction (35 REGISTER) completed 47<SIP0>: No opaque in authentication 48<SIP0>: Adding authentication information 49<SIP36>: Send REGISTER Request Sending SIP PDU to ( 188.134.000.243:5060 ) from 5060 REGISTER sip:188.134.000.243 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 36 REGISTER Date: Tue, 27 Sep 2011 14:56:28 GMT User-Agent: AddPac SIP Gateway Authorization: Digest username="14924", realm="voip.interzet.ru", nonce="661f5f4c", uri="sip:188.134.000.243", response="c44ef2db3c4a29b0f58f3153b0201b06", algorithm=MD5 Contact: <sip:14924@192.168.1.20>;expires=60 Expires: 60 Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 188.134.0.243:5060 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436;received=188.134.30.161 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 36 REGISTER User-Agent: InterZet PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:14924@188.134.0.243> Content-Length: 0 50<SIP36>: Receive 100 Trying 51<SIP36>: Transaction (36 REGISTER) proceeding Received SIP PDU from ( 188.134.0.243:5060 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436;received=188.134.30.161 From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4 To: sip:14924@188.134.000.243;tag=as217b6d45 Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20 CSeq: 36 REGISTER User-Agent: InterZet PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: <sip:14924@192.168.1.20>;expires=60 Date: Tue, 27 Sep 2011 10:56:21 GMT Content-Length: 0 52<SIP36>: Receive 200 OK 53<SIP36>: Transaction (36 REGISTER) completed 54<SIP35>: Set Terminated Success for 35 REGISTER 55<SIP36>: Set Terminated Success for 36 REGISTER [848.095] VM(0/0/0) vmOnHook [848.145] VM(0/0/0) vmTmoOnHook [848.195] VM(0/0/0) vmTmoOnHook [848.245] VM(0/0/0) vmTmoOnHook [848.295] VM(0/0/0) vmTmoOnHook [848.345] VM(0/0/0) vmTmoOnHook [848.395] VM(0/0/0) vmTmoOnHook [848.445] VM(0/0/0) vmTmoOnHook [848.495] VM(0/0/0) vmTmoOnHook [848.545] VM(0/0/0) vmTmoOnHook [848.595] VM(0/0/0) vmTmoOnHook [848.645] VM(0/0/0) vmTmoOnHook [848.695] VM(0/0/0) vmTmoOnHook [848.745] VM(0/0/0) vmTmoOnHook [848.795] VM(0/0/0) vmTmoOnHook [848.795] VM(0/0/0) Rx OnHook [848.795] VM(0/0/0) vopp idle [848.795] VM(0/0/0) Tx DISCONN_CNF 56<CEP000000>: Disconnected(16) at Disconnecting Received SIP PDU from ( 188.134.0.243:5060 ) OPTIONS sip:14924@192.168.1.20 SIP/2.0 Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK3cd655ea;rport From: "asterisk" <sip:asterisk@voip.interzet.ru>;tag=as654a36f6 To: <sip:14924@192.168.1.20> Contact: <sip:asterisk@188.134.0.243> Call-ID: 419ee2010136040e48f1c6c5626551bc@voip.interzet.ru CSeq: 102 OPTIONS User-Agent: InterZet PBX Max-Forwards: 70 Date: Tue, 27 Sep 2011 10:56:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK3cd655ea;rport From: "asterisk" <sip:asterisk@voip.interzet.ru>;tag=as654a36f6 To: <sip:14924@192.168.1.20> Call-ID: 419ee2010136040e48f1c6c5626551bc@voip.interzet.ru CSeq: 102 OPTIONS User-Agent: AddPac SIP Gateway Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Length: 0 |
Автор: | craft [ 28 сен 2011, 09:32 ] |
Заголовок сообщения: | Re: Факсы ходят только в одну сторону |
sip-server 188.134.000.243 - это сервер провайдера? Если да, то мучать его |
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