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| IP90 входящие вызовы с AS5350 http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2365 |
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| Автор: | Vov_ [ 28 фев 2011, 05:10 ] |
| Заголовок сообщения: | IP90 входящие вызовы с AS5350 |
Здравствуйте, подскажите пожл, есть такая связка IP90 и AS5303, связь по сип, звонок с ип90 на ас5350 проходит, а вот наоборот появилась проблема, сам звонок проходит, но при поднятии трубки ип90 проходит отбой с Remote:CallClear. конфиг ас5350 dial-peer voice 396877 voip huntstop destination-pattern 396877 voice-class codec 2 session protocol sipv2 session target ipv4:ХХ.ХХ.ХХ.ХХ dtmf-relay rtp-nte sip-notify sip-kpml fax-relay ecm disable fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none no vad конфиг ИП90: Current configuration: ! version 8.51.001 ! hostname Anton clock timezone URALS 5 ! username root password router administrator ! ! interface Loopback0 ip address 127.0.0.1 255.0.0.0 ! interface FastEthernet0/0 ip address ХХ.ХХ.ХХ.ХХ ip nat outside speed auto no qos-control ! interface FastEthernet0/1 ip address 192.168.10.1 255.255.255.0 ip nat inside speed auto no qos-control ! ip route 0.0.0.0 0.0.0.0 ХХХХХХХ ! access-list 100 permit ip 192.168.10.0 0.0.0.255 any ! ip nat inside source list 100 interface FastEthernet0/0 overload ! snmp name IP90_G2 ! ip tcp keep-alive count 5 ip tcp keep-alive idle 60 ip tcp keep-alive interval 5 ! ftp server http server ! ! ! IP PHONE OSD configuration. ! osd language russian network signaling sip network sscp disable phone save-mode always phone lcd-type graphic phone ring-type 6 phone volume ring 5 phone volume input 5 phone volume output 5 phone volume micbooster disable phone auto-hook-on disable phone display-name TRS phone dnd-mode silence phone pbx-mode general phone hook-mode digitwithdirect phone auto-answer 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 disable 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 phone noanswer-sound off phone noanswer-sound notify interval 30 phone emergency-number 1 112 phone emergency-number 2 119 phone emergency-number 3 911 phone emergency-number 4 999 phone contact-profile id 0 version 0 phone recording disable ! ! SSCP configuration.! ! ! ! SSCP Static CM List sscp ! ! SSCP Dynamic CM List sscp ! ! sscp call-manager broadcast port 8855 logger disable logger level info ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol t38 redundancy 0 fax rate disable timeout tmohdt 300 call-barring unconfigured-ip-address voip-inbound-call-barring enable ! ! ! Voice port configuration. ! ! SPEECH voice-port 0/0 fax-early-detect ! ! ! N/A voice-port 0/1 ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 396877 port 0/0 huntstop recording all ! dial-peer voice 1 pots destination-pattern #8 port 0/1 huntstop recording all ! ! ! ! Voip peer configuration. ! dial-peer voice 100 voip destination-pattern T session target ip УУ.УУ.УУ.УУ codec g711ulaw voice-class codec 0 no vad dtmf-relay rtp-2833 translate-outgoing called-number 1 fax protocol t38 redundancy 0 fax rate 9600 recording all ! dial-peer voice 101 voip destination-pattern 8T session target ip ZZ.ZZ.ZZ.ZZ codec g711ulaw voice-class codec 0 no vad dtmf-relay rtp-2833 fax protocol t38 redundancy 0 fax rate 9600 recording all ! dial-peer voice 1001 voip destination-pattern T session target sip-server voice-class codec 0 no vad dtmf-relay rtp-2833 huntstop shutdown recording all ! dial-peer voice 1003 voip voice-class codec 0 no vad dtmf-relay rtp-2833 shutdown recording all ! dial-peer voice 1004 voip voice-class codec 0 no vad dtmf-relay rtp-2833 shutdown recording all ! ! ! dial-peer ipaddr-prefix n dial-peer call-hold n dial-peer call-transfer n ! ! ! ! Gateway configuration. ! gateway ! ! ! ! Recording configuration. ! recording direction all ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729 codec preference 4 g7231r63 codec preference 5 g726r32 ! ! ! ! Translation Rule configuration. ! translation-rule 1 rule 1 1 9991 rule 2 2 9992 rule 3 3 9993 rule 4 4 9994 rule 5 5 9995 rule 6 6 9996 rule 7 7 9997 rule 8 0 9990 rule 9 9 9999 ! ! ! ! SIP UA configuration. ! sip-ua no fault-tolerance no rport call-transfer-mode attended recording-info-notify ! ! ! SMS UA configuration. ! sms-ua check-to-header enable ! ! ! Tones ! ! ! ! line console ! line vty ! ! sms quota 30 ! ! ! PS Client configuration. ! psclient service disable retry_timer 10000 alive_timer 15000 ! ! end |
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| Автор: | genal [ 28 фев 2011, 10:29 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
Дебаг можете показать? |
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| Автор: | Vov_ [ 01 мар 2011, 03:11 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
Anton# Received SIP PDU from ( (ИП ас5350)ХХХХ:53702 ) INVITE sip:396877@(ИП ип90)УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDPХХХХ:5060;branch=z9hG4bK7B014C1 Remote-Party-ID: <sip:83512396801@УУУУ>;party=calling;screen=yes;privacy=off From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ> Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 754866311-1122374112-2323972105-3893905184 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 65 Timestamp: 1298948118 Contact: <sip:83512396801@ХХХХ:5060> Call-Info: <sip:ХХХХ:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Expires: 180 Allow-Events: kpml, telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 309 v=0 o=CiscoSystemsSIP-GW-UserAgent 7700 6269 IN IP4 ХХХХ s=SIP Call c=IN IP4 ХХХХ t=0 0 m=audio 17544 RTP/AVP 0 8 18 101 c=IN IP4 ХХХХ a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Sending SIP PDU to ( ХХХХ:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ> Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 101 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 537 <Call 29> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1262387150) **** 538 <SIP 29> : Receive INVITE Request 539 <NetCon 29> : Found inbound voip peer(100) result(2) peer->fixedPatternSize(0) mostMatchingSize(-1) 540 <NetCon 29> : Found inbound voip peer by dest-pattern id(100) 541 <NetCon 29> : Found inbound voip peer(101) result(2) peer->fixedPatternSize(1) mostMatchingSize(0) 542 <NetCon 29> : Found inbound voip peer by dest-pattern id(101) 543 <NetCon 29> : Found inbound voip peer(1001) result(2) peer->fixedPatternSize(0) mostMatchingSize(1) 544 <Call 29> : From Net - calledParty(396877) callingParty(83512396801) 545 <Call 29> : MatchedPerfect 546 <Call 29> : MatchAllProcess After Sorted <0> id(0) dest(396877) prefer(0) selected(17) 547 <Call 29> : Initiate callee with dial-peer(396877) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 548 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(83512396801), callerPort(ffffffff) type(SPEECH) 549 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(29) CallState(Ringing:4) Softkey(Ringing:7) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE) 550 <CCA 0> : Call Received from Net : Address(ХХХХ) Number(83512396801) Name(83512396801) 551 <SIP29> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 552 <SIP 29> : SetLocalAudioFormats : myVoipPeer(101) is not NULL, codec(1) 553 <PhonePlay 29> : Audio Count(1) 554 <PhonePlay 29> : rtpSessionId(1) Second Audio Port(-1) 555 <SIP 29> : SetAlerting 556 <SIP 29> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 557 <SIP 29> : SetLocalAudioFormats : myVoipPeer(101) is not NULL, codec(1) 558 <SIP 29> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( ХХХХ:5060 ) from 5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 101 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:396877@УУУУ RSeq: 169898 Require: 100rel Content-Type: application/sdp Content-Length: 235 v=0 o=396877 1262387150 1262387150 IN IP4 УУУУ s=AddPac Gateway SDP c=IN IP4 УУУУ t=1262387150 0 m=audio 23044 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Received SIP PDU from ( ХХХХ:53702 ) PRACK sip:396877@УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B11691 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 102 PRACK RAck: 169898 101 INVITE Max-Forwards: 65 Content-Length: 0 559 <SIP 29> : Receive PRACK Request Sending SIP PDU to ( ХХХХ:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B11691 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 102 PRACK User-Agent: AddPac SIP Gateway Content-Length: 0 [ohx] [OSD->CCC] REQ (CallConnect) (voip_commands.cxx/24041) [ohx] [OSD->CCC] REQ(1) (voip_commands.cxx/24042) 560 <CCA 0> : Call Connect Request from HANDSET 561 <Call 29> : Connected from(0) 562 <CEP 000000> : StopSignal 563 <SIP 29> : SetConnected Sending SIP PDU to ( ХХХХ:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 101 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:396877@УУУУ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Length: 0 564 <CCA 0> : Call Connected from Device (cn=29) CallState(Connected:5) Softkey(Connected:5) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE) Received SIP PDU from ( ХХХХ:53702 ) ACK sip:396877@УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B2A6B From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ Max-Forwards: 65 CSeq: 101 ACK Allow-Events: kpml, telephone-event Content-Length: 0 565 <SIP 29> : ACK received 566 <SIP 29> : Receive ACK Request 567 <SIP 29> : Set Terminated Success for 101 INVITE Received SIP PDU from ( ХХХХ:53702 ) BYE sip:396877@УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 65 Timestamp: 1298948121 CSeq: 103 BYE Reason: Q.850;cause=96 Content-Length: 0 568 <SIP 29> : Receive BYE Request Sending SIP PDU to ( ХХХХ:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ CSeq: 103 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 569 <SIP 29> : ReleaseWithNothing 570 <Call 29> : Terminated from(fffffffe) this(Remote:CallClear) before((null)) forced(0) time(1262387153) [ohx] CCC(id:0x0) --> SetASStatus(OnHooKStatus -> OnHooKStatus) : (voip_call.cxx,13276) 571 <CEP 000000> : SetAdditional Service Status fr(OnHooKStatus) to(OnHooKStatus) CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE) 572 <CEP 000000> : DisconnectCall at Busy 573 <CEP 000000> : StopSignal CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0xffffffff) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE) 574 <CEP 000000> : Disconnect (0) 575 <CEP 000000> : Disconnect (0) 576 <CCA 0> : Call Disconnected Error : Current call is not assigned callNumber(29), [0][0][0][0]:[0][0][0][0] CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0xffffffff) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE) 577 <NetEP 29> : Call FROM <83512396801> terminated reason(Remote:CallClear) 578 <CEP 000000> : Disconnected(16) at Disconnecting [ohx] [OSD->CCC] REQ (CallDisconn) (voip_commands.cxx/24041) [ohx] [OSD->CCC] REQ(2) (voip_commands.cxx/24042) 579 <CCA 0> : Call Disconnect Request from HANDSET callNumber(-1) 580 <CCA 0> : Call Disconnect Error : Current call is not assigned callNumber(-1) [0][0][0][0] |
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| Автор: | Yuri [ 02 мар 2011, 13:37 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
Почему-то bye c AS5350 приходит c кодом 96: Received SIP PDU from ( ХХХХ:53702 ) BYE sip:396877@УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130 From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 65 Timestamp: 1298948121 CSeq: 103 BYE Reason: Q.850;cause=96 Content-Length: 0 |
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| Автор: | Vov_ [ 03 мар 2011, 03:17 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
спасибо, но стал смотреть дальше и увидел вот что Received SIP PDU from ( ХХХХ:53702 ) ACK sip:396877@УУУУ:5060 SIP/2.0 Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B2A6B From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE To: <sip:396877@УУУУ>;tag=ce4b382ba4 Date: Tue, 01 Mar 2011 02:55:18 GMT Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ Max-Forwards: 65 CSeq: 101 ACK Allow-Events: kpml, telephone-event Content-Length: 0 как я понимаю это сообщение об авторизации, и как-то кто-то должен авторизироваться, хотя на циске все разрешения прописаны только для ип адресов. |
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| Автор: | Yuri [ 04 мар 2011, 07:15 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
Пакеты ACK используются для подтверждений приема других пакетов. У Вас в схеме авторизация не используется. |
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| Автор: | Vov_ [ 05 мар 2011, 04:18 ] |
| Заголовок сообщения: | Re: IP90 входящие вызовы с AS5350 |
нет авторизации нет, есть ацл листы на воипный трафик и все. |
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