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AP1100f. номер исходящего звонка. http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2553 |
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Автор: | lem [ 15 дек 2011, 05:44 ] |
Заголовок сообщения: | AP1100f. номер исходящего звонка. |
Доброе утро. Есть AP1000f 4 FXS. Настрое на Астериск по сип. На 0/0-0/3 порты прописаны учетки, каждая со своим номером, при звонке на шлюз по каждому из номеров - звонок проходит и связь работает, но если звонить с любого из номеров то звонок определяется номером с порта 0/0. При этом, если позвонить с аккаунта на астериске то нормер определится верно. В чем может быть проблема? Код: AP1000# sh run ! version 8.30K ! 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 ! ! ip-share enable ip-share interface net-side ether0.0 ip-share interface local-side ether1.0 ! interface ether0.0 ip address dhcp ! interface ether1.0 no ip address ! snmp name AP1000 ! no arp reset ! ! ! ! ! ! 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 1 pots destination-pattern ***7969 port 0/0 user-name 6*****a user-password ******** ! dial-peer voice 2 pots destination-pattern ***7989 port 0/1 user-name c*****0 user-password ******** ! ! ! ! Voip peer configuration. ! dial-peer voice 100 voip destination-pattern T session target sip-server session protocol sip codec g729 dtmf-relay rtp-2833 no vad ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.10.21.32.5 no ignore-msg-from-other-gk ! ! ! SIP UA configuration. ! sip-ua user-register sip-server 46.2*.*.* register e164 ! ! ! MGCP configuration. ! mgcp codec g711ulaw vad ! ! ! Tones ! ! ! ! Лог исходящего звонка deb voip call: Код: AP1000# deb voip call AP1000# 1 <CEP 000100> : Call Received 2 <CEP 000100> : Call Initiated : calledNumber() crv(0) total(0) 3 <Call 8> : ****************** Call Created status(InitiatedByFXS ) ******************* 4 <CEP 000100> : Calling number(***7989) 5 <CEP 000100> : Call id(f4a8a54e-5815-6576-8012-0002a404f92a) callNum( 8) 6 <Call 8> : Digit(8) at InitiatedByFXS 7 <Call 8> : MatchedAll 8 <Call 8> : Digit(9) at CalleeDeterminedWaitDigit 9 <Call 8> : MatchedAll 10 <Call 8> : Digit(2) at CalleeDeterminedWaitDigit 11 <Call 8> : MatchedAll 12 <Call 8> : Digit(7) at CalleeDeterminedWaitDigit 13 <Call 8> : MatchedAll 14 <Call 8> : Digit(2) at CalleeDeterminedWaitDigit 15 <Call 8> : MatchedAll 16 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 17 <Call 8> : MatchedAll 18 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 19 <Call 8> : MatchedAll 20 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 21 <Call 8> : MatchedAll 22 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 23 <Call 8> : MatchedAll 24 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 25 <Call 8> : MatchedAll 26 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit 27 <Call 8> : MatchedAll 28 <Time 8> : Inter digit timer timeout. 29 <Call 8> : Digit(#) at CalleeDeterminedWaitDigit 30 <Call 8> : MatchAllProcess After Sorted <0> id(100) dest(T) prefer(0) selected(2) 31 <Call 8> : Initiate callee with dial-peer(T) status(CalleeDetermi nedAll) id(f4a8a54e-5815-6576-8012-0002a404f92a) 32 <NetEP 8> : InitiateOutCall: calledNum(89272******) callingNum(205 7989) target(sip-server) 33 <NetEP 8> : DoCall: calledAddr(sip:89272******@46.2*.*.*:5060) ca llingAddr(2057989) 34 <SIP 0> : No authentication information available 35 <SIP 8> : Send INVITE Request 36 <SIP 8> : Receive 100 Trying 37 <SIP 8> : Transaction (6 INVITE) proceeding 38 <SIP 8> : Receive 200 OK 39 <SIP 8> : Get SIP Audio MediaFormat : 18 40 <Call 8> : Connected from(fffffffe) 41 <NetEP 8> : Call with sip:89272******@46.2*.*.* established 42 <SIP 8> : Received INVITE OK response 43 <SIP 8> : Send ACK Request 44 <SIP 8> : Check Event Relation 45 <SIP 8> : Set Terminated Success for 6 INVITE 46 <Time 0> : SIP_TREGISTER timer timeout. 47 <SIP 0> : Adding authentication information 48 <SIP 4735> : Send REGISTER Request 49 <SIP 4735> : Receive 401 Unauthorized 50 <SIP 4735> : Transaction (4735 REGISTER) completed 51 <SIP 0> : No opaque in authentication 52 <SIP 0> : Adding authentication information 53 <SIP 4736> : Send REGISTER Request 54 <SIP 4736> : Receive 200 OK 55 <SIP 4736> : Transaction (4736 REGISTER) completed 56 <SIP 4736> : Send SNMP TRAP (success with code - 200) previous succ ess 57 <SIP 0> : Adding authentication information 58 <SIP 4737> : Send REGISTER Request 59 <SIP 4737> : Receive 401 Unauthorized 60 <SIP 4737> : Transaction (4737 REGISTER) completed 61 <SIP 0> : No opaque in authentication 62 <SIP 0> : Adding authentication information 63 <SIP 4738> : Send REGISTER Request 64 <SIP 4738> : Receive 200 OK 65 <SIP 4738> : Transaction (4738 REGISTER) completed 66 <SIP 4738> : Send SNMP TRAP (success with code - 200) previous succ ess 67 <SIP 4735> : Set Terminated Success for 4735 REGISTER 68 <SIP 4736> : Set Terminated Success for 4736 REGISTER 69 <SIP 4737> : Set Terminated Success for 4737 REGISTER 70 <SIP 4738> : Set Terminated Success for 4738 REGISTER 71 <CEP 000100> : Disconnected(16) at Busy 72 <Call 8> : Terminated from(100) this(Local:CallClear) before(NUL L) forced(0) 73 <CEP 000100> : DisconnectCall at Idle 74 <SIP 8> : ReleaseWithBYE 75 <SIP 8> : Send BYE Request 76 <NetEP 8> : Call TO <sip:89272******@46.2*.*.*> terminated reason (Local:CallClear) 77 <SIP 8> : Receive 200 OK 78 <SIP 8> : Transaction (7 BYE) completed 79 <SIP 8> : Set Terminated Success for 7 BYE Звонок пришел с номера ***7969 |
Автор: | genal [ 26 дек 2011, 13:02 ] |
Заголовок сообщения: | Re: AP1100f. номер исходящего звонка. |
debug voip sip можно посмотреть? |
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