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Addpac gs-1002 сбрасывает звонок через GSM порт после 120 секунд разговора [закрыт]

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Здравствуйте Addpac gs-1002 после перепрошивки стал сбрасывать звонок через GSM порт после 120 секунд разговора и не работает время определения звонка, и также не определяется кодек,Через FXO все идет на ура

Измененные настройки Addpac gs-1002

 
! ! APOS(tm) configuration saved from vty ! 2014/08/21 23:42:35 ! version 8.51.010 ! hostname GS1002 clock timezone Chita 10 ! username root password router administrator username guest password guest user ! ! script ntpdate default resynchronize 1 0 server ip us.pool.ntp.org ! interface Loopback0 ip address 127.0.0.1 255.0.0.0 ! interface FastEthernet0/0 ip address 192.168.211.32 255.255.255.0 speed auto no qos-control ! interface FastEthernet0/1 no ip address speed auto no qos-control ! interface FastEthernet0/1:1 ip address 192.168.10.1 255.255.255.0 ! ip route 0.0.0.0 0.0.0.0 192.168.211.1 10 ! ! ! ! ftp server http server ! logging command logging event 4-warning logging on ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip protocol sip dtmf-relay rfc-2833 fax protocol t38 redundancy 0 fax rate 9600 h323 call start fast h323 call tunnel enable timeout tinit 15 timeout tidt 5 static-jitter-buffer 35 ignore-dtmf-abcd-tone no call-barring unconfigured-ip-address no voip-inbound-call-barring enable ! ! ! Voice port configuration. ! ! GSM voice-port 0/0 connection plar 201 caller-id enable caller-id name disable ! ! ! GSM voice-port 0/1 connection plar 202 caller-id enable caller-id name disable ! ! ! FXO voice-port 0/2 connection plar 203 ring detect-timeout 80 caller-id enable caller-id name disable ! ! ! FXO voice-port 0/3 connection plar 204 ring detect-timeout 80 caller-id enable caller-id name disable ! ! ! ! ! service port group configuration. ! ! ! ! Pots peer configuration. ! dial-peer voice 900 pots destination-pattern 01T port 0/0 no register e164 translate-outgoing called-number 900 ! dial-peer voice 901 pots destination-pattern 02T port 0/1 no register e164 translate-outgoing called-number 901 ! dial-peer voice 902 pots destination-pattern 03T port 0/2 no register e164 translate-outgoing called-number 902 ! dial-peer voice 903 pots destination-pattern 04T port 0/3 no register e164 translate-outgoing called-number 903 ! ! ! ! Voip peer configuration. ! dial-peer voice 1 voip destination-pattern 201 session target ip 192.168.211.3
session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 no sid ! dial-peer voice 2 voip destination-pattern 202 session target ip 192.168.211.3
session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 no sid ! dial-peer voice 3 voip destination-pattern 203 session target ip 192.168.211.3
session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 no sid ! dial-peer voice 4 voip destination-pattern 204 session target ip 192.168.211.3
session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 no sid ! ! ! ! ! ! gatekeeper ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.211.32 no ignore-msg-from-other-gk shutdown ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711alaw codec preference 2 g711alaw codec preference 3 g729 ! ! ! ! Translation Rule configuration. ! translation-rule 900 rule 0 01T T
! translation-rule 901 rule 0 02T T
! translation-rule 902 rule 1 03T T
! translation-rule 903 rule 0 03T T
! ! ! ! SIP UA configuration. ! sip-ua sip-server 192.168.211.3 5060 126 remote-party-id ! ! ! Tones ! ! ! SMS delivery configuration ! sms-delivery ! ! ! ! ! line console ! line vty ! mobile dev-restart-by-unreg 300 no mobile dev-restart-by-unknown-error mobile cell-monitor 30 ! mobile 0/0 gsm sms-language utf8 ! mobile 0/1 gsm sms-language utf8 !

Дебаг с TRIXBOX (астериска)

не рабочий Звонок на GSM

asterisk -r
rtp debug on
-- Registered SIP '105' at 192.168.211.101 port 58056
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Executing [89993338888@from-internal:1] Macro("SIP/105-0000090e", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/105-0000090e", "AMPUSER=105") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/105-0000090e", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/105-0000090e", "1?Set(REALCALLERIDNUM=105)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/105-0000090e", "AMPUSER=105") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/105-0000090e", "AMPUSERCIDNAME=Maltsev IS SoftF") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/105-0000090e", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/105-0000090e", "AMPUSERCID=105") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/105-0000090e", "CALLERID(all)="Maltsev IS SoftF" <105>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/105-0000090e", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/105-0000090e", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/105-0000090e", "Using CallerID "Maltsev IS SoftF" <105>") in new stack
    -- Executing [89993338888@from-internal:2] Set("SIP/105-0000090e", "NODEST=") in new stack
    -- Executing [89993338888@from-internal:3] Macro("SIP/105-0000090e", "record-enable,105,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/105-0000090e", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/105-0000090e", "recordingcheck,20140825-065642,1408949802.2318") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    -- astgetsrv: SRV lookup for 'sip.UDP.multifon.ru' mapped to host sbc.multifon.ru, port 5060
 recordingcheck,20140825-065642,1408949802.2318: Outbound recording not enabled
    -- <sip 105-0000090e="">AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/105-0000090e", "") in new stack
    -- Executing [89993338888@from-internal:4] Macro("SIP/105-0000090e", "dialout-trunk,12,89993338888,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/105-0000090e", "DIALTRUNK=12") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/105-0000090e", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/105-0000090e", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/105-0000090e", "DIALNUMBER=89993338888") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/105-0000090e", "DIALTRUNKOPTIONS=trTw") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/105-0000090e", "OUTBOUNDGROUP=OUT12") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/105-0000090e", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/105-0000090e", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/105-0000090e", "DIALTRUNKOPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/105-0000090e", "outbound-callerid,12") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/105-0000090e", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/105-0000090e", "0?Set(REALCALLERIDNUM=105)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/105-0000090e", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/105-0000090e", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/105-0000090e", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/105-0000090e", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/105-0000090e", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/105-0000090e", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/105-0000090e", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/105-0000090e", "0?Set(CALLERPRES()=prohibpassed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/105-0000090e", "1?AGI(fixlocalprefix)") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

fixlocalprefix: Using pattern 02+. == fixlocalprefix: Dialpattern 02+. matched. 89993338888 -> 0289993338888 -- <sip 105-0000090e="">AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:13] Set("SIP/105-0000090e", "OUTNUM=0289993338888") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/105-0000090e", "custom=SIP/01") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/105-0000090e", "0?Set(DIALTRUNKOPTIONS=M(setmusic^))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/105-0000090e", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/105-0000090e", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/105-0000090e", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/105-0000090e", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/105-0000090e", "SIP/01/0289993338888,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Called 01/0289993338888 -- SIP/01-0000090f is making progress passing it to SIP/105-0000090e ; Начался разоговор (слышимость в обе стороны) который длится 120 секунд затем синнал занято и: -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.211.32 -- SIP/01-0000090f is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] Goto("SIP/105-0000090e", "s-CONGESTION,1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/105-0000090e", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,3) -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/105-0000090e", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing [89993338888@from-internal:5] Macro("SIP/105-0000090e", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Playback("SIP/105-0000090e", "all-circuits-busy-now,noanswer") in new stack -- <sip 105-0000090e=""> Playing 'all-circuits-busy-now.alaw' (language 'ru') -- Executing [s@macro-outisbusy:2] Playback("SIP/105-0000090e", "pls-try-call-later,noanswer") in new stack -- <sip 105-0000090e=""> Playing 'pls-try-call-later.alaw' (language 'ru') -- Executing [s@macro-outisbusy:3] Macro("SIP/105-0000090e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/105-0000090e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/105-0000090e", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/105-0000090e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/105-0000090e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/105-0000090e' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/105-0000090e' in macro 'outisbusy' == Spawn extension (from-internal, 89993338888, 5) exited non-zero on 'SIP/105-0000090e' -- Executing [h@from-internal:1] Macro("SIP/105-0000090e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/105-0000090e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/105-0000090e", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/105-0000090e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/105-0000090e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/105-0000090e' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/105-0000090e'

;Рабочий Звонок на FXO

== Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Executing [226729@from-internal:1] Macro("SIP/105-00000910", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/105-00000910", "AMPUSER=105") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/105-00000910", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/105-00000910", "1?Set(REALCALLERIDNUM=105)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/105-00000910", "AMPUSER=105") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/105-00000910", "AMPUSERCIDNAME=Maltsev IS SoftF") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/105-00000910", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/105-00000910", "AMPUSERCID=105") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/105-00000910", "CALLERID(all)="Maltsev IS SoftF" <105>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/105-00000910", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/105-00000910", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/105-00000910", "Using CallerID "Maltsev IS SoftF" <105>") in new stack
    -- Executing [226729@from-internal:2] Set("SIP/105-00000910", "NODEST=") in new stack
    -- Executing [226729@from-internal:3] Macro("SIP/105-00000910", "record-enable,105,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/105-00000910", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/105-00000910", "recordingcheck,20140825-070812,1408950492.2320") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20140825-070812,1408950492.2320: Outbound recording not enabled
    -- <sip 105-00000910="">AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/105-00000910", "") in new stack
    -- Executing [226729@from-internal:4] Macro("SIP/105-00000910", "dialout-trunk,10,226729,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/105-00000910", "DIALTRUNK=10") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/105-00000910", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/105-00000910", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/105-00000910", "DIALNUMBER=226729") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/105-00000910", "DIALTRUNKOPTIONS=trTw") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/105-00000910", "OUTBOUNDGROUP=OUT10") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/105-00000910", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/105-00000910", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/105-00000910", "DIALTRUNKOPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/105-00000910", "outbound-callerid,10") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/105-00000910", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/105-00000910", "0?Set(REALCALLERIDNUM=105)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/105-00000910", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/105-00000910", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/105-00000910", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/105-00000910", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/105-00000910", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/105-00000910", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/105-00000910", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/105-00000910", "0?Set(CALLERPRES()=prohibpassed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/105-00000910", "1?AGI(fixlocalprefix)") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

fixlocalprefix: Using pattern 04+. == fixlocalprefix: Dialpattern 04+. matched. 226729 -> 04226729 -- <sip 105-00000910="">AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:13] Set("SIP/105-00000910", "OUTNUM=04226729") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/105-00000910", "custom=SIP/357093") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/105-00000910", "0?Set(DIALTRUNKOPTIONS=M(setmusic^))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/105-00000910", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/105-00000910", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/105-00000910", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/105-00000910", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/105-00000910", "SIP/357093/04226729,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Called 357093/04226729 -- SIP/357093-00000911 is making progress passing it to SIP/105-00000910 -- SIP/357093-00000911 answered SIP/105-00000910 -- Executing [h@macro-dialout-trunk:1] Macro("SIP/105-00000910", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/105-00000910", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/105-00000910", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/105-00000910", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/105-00000910", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/105-00000910' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/105-00000910' in macro 'dialout-trunk' == Spawn extension (from-internal, 226729, 4) exited non-zero on 'SIP/105-00000910'

Дебаги Addpac

Исходящий на GSM

GS1002# terminal monitor
GS1002# debug voip call
GS1002# 1       <call 42="">     : *  Call Created status(InitiatedByNet) ver(8.51:2011-02-06-00-00) time(1408670039) ***
2       <sip 42="">     : Receive INVITE Request
3       <netcon 42="">     : Found inbound voip peer by IP address id(1)
4       <call 42="">     : From Net - calledParty(0289990008888) callingParty(105)
5       <call 42="">     : MatchedAll
6       <call 42="">     : MatchAllProcess After Sorted
                          <0>  id(901) dest(02T) prefer(0) selected(12)
7       <call 42="">     : Initiate callee with dial-peer(02T) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8       <cep 000100=""> : InitiateOutCall :  calledNum(0289990008888), callingNum(105), callerPort(ffffffff) type(GSM)
9       <cep 000100=""> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(42)
10      <sip 42="">     : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
11      <sip 42="">     : SetLocalAudioFormats : myVoipPeer(1) is not NULL, voiceCodecClass(0)
12      <phoneplay 42="">     : Audio Count(1)
13      <phoneplay 42="">     : rtpSessionId(1) Second Audio Port(-1)
14      <sip 42="">     : SetAlerting
15      <call 42="">     : PreConnected from(100)
16      <sip 42="">     : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
17      <sip 42="">     : SetLocalAudioFormats : myVoipPeer(1) is not NULL, voiceCodecClass(0)
18      <sip 42="">     : Add Local Audio MediaFormat : 8
19      <time 42="">     : Call Forwarding No Answer timer timeout.

;Разговор начался (слышимость в обе стороны), сигнал занято через 120 секунд

20 <cep 000100=""> : Disconnected(16) at Busy 21 <call 42=""> : Terminated from(100) this(Local:CallClear) before(NULL) forced(0) time(1408670160) 22 <netep 42=""> : Call FROM <maltsev is="" softf=""> terminated reason(Local:CallClear) 23 <cep 000100=""> : DisconnectCall at Idle 24 <sip 42=""> : Receive ACK Request 25 <sip 42=""> : Set Terminated Success for 102 INVITE

Исходящий на FXO

GS1002# 
26      <call 43="">     : *  Call Created status(InitiatedByNet) ve                             r(8.51:2011-02-06-00-00) time(1408670420) *** 
27      <sip 43="">     : Receive INVITE Request 
28      <netcon 43="">     : Found inbound voip peer by IP address id(1)
29      <call 43="">     : From Net - calledParty(04226729) callingParty(101)
30      <call 43="">     : MatchedAll
31      <call 43="">     : MatchAllProcess After Sorted
                          <0>  id(903) dest(04T) prefer(0) selected(4)
32      <call 43="">     : Initiate callee with dial-peer(04T) status(CalleeDeter                             minedAll) id(00000000-0000-0000-0000-000000000000)
33      <cep 000300=""> : InitiateOutCall :  calledNum(226729), callingNum(101),                              callerPort(ffffffff) type(FXO)
34      <cep 000300=""> : Outbound call to CEP callId(00000000-0000-0000-0000-00                             0000000000) callNum(43)
35      <sip 43="">     : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE                             )
36      <sip 43="">     : SetLocalAudioFormats : myVoipPeer(1) is not NULL, voic                             eCodecClass(0)
37      <phoneplay 43="">     : Audio Count(1)
38      <phoneplay 43="">     : rtpSessionId(1) Second Audio Port(-1)
39      <sip 43="">     : SetAlerting
40      <call 43="">     : PreConnected from(300)
41      <sip 43="">     : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE                             )
42      <sip 43="">     : SetLocalAudioFormats : myVoipPeer(1) is not NULL, voic                             eCodecClass(0)
43      <sip 43="">     : Add Local Audio MediaFormat : 8
44      <call 43="">     : Connected from(300)
45      <sip 43="">     : SetConnected
46      <sip 43="">     : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE                             )
47      <sip 43="">     : SetLocalAudioFormats : myVoipPeer(1) is not NULL, voic                             eCodecClass(0)
48      <sip 43="">     : Add Local Audio MediaFormat : 8
49      <sip 43="">     : ACK received
50      <sip 43="">     : Receive ACK Request
51      <sip 43="">     : Set Terminated Success for 102 INVITE
52      <sip 43="">     : Receive BYE Request
53      <sip 43="">     : ReleaseWithNothing
54      <call 43="">     : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0) time(1408670517)
55      <cep 000300=""> : DisconnectCall at Busy
56      <cep 000300=""> : StopSignal
57      <cep 000300=""> : Disconnect (0)
58      <netep 43="">     : Call FROM <maltsev is=""> terminated reason(Remote:CallClear)
59      <cep 000300=""> : Disconnected(16) at Disconnecting
60      <cep 000300=""> : Call Received
61      <cep 000300=""> : Disconnected(16) at Busy

удалить переоткрыть спам изменить тег редактировать

спросил 2014-08-21 07:48:28 +0400

mivan Gravatar mivan
1 2 1

обновил 2014-08-25 11:50:38 +0400

Comments

вот мануал - http://awsswa.livejournal.com/22887.html

awsswa ( 2014-08-21 08:02:59 +0400 )редактировать

awsswa спасибо настроил как в мануале все заработало, а где собака была зарыта в каком параметре?

mivan ( 2014-08-21 17:48:43 +0400 )редактировать

ну вот возьмите и найдите. почему вы думаете что за васэто ктото должен далать?

meral ( 2014-08-21 19:18:48 +0400 )редактировать

Рано радовался, не работает. При исходящих с трибокса на GSM звонки уходят на нужный транк но на шлюзе не определяется время начала разговора и не идет тайминг разговора, хотя разговор идет и слышимость в обе стороны, и я думаю что из-за этого идет обрыв разговора на 120 секунде. Входящие идут нормально, FXO в обе стороны хорошо Хелп

mivan ( 2014-08-21 20:29:16 +0400 )редактировать

добро пожаловать в мир где нужны специализированные знания. почемуто машину вы себе не собираете. читайте документацию.

meral ( 2014-08-21 21:03:08 +0400 )редактировать

meral подобные сайты и созданы для того, чтобы получить специализированные знания

mivan ( 2014-08-22 03:58:54 +0400 )редактировать

в астериск в консоле - rtp debug on - и смотрите куда у вас голос бегает - очень похоже на проблемы с маршрутизацией или firewall

awsswa ( 2014-08-22 07:08:13 +0400 )редактировать

не по addpac. для получения знание по addpac надо читать документацию и его родной форум. да и вообще fxo не самые простые железки для настройки. не говоря про addpac.

meral ( 2014-08-22 12:55:04 +0400 )редактировать

Сделал debug и в астериске и в Addpack, мне кажется что проблема в Addpac, но из-за чего не пойму.

mivan ( 2014-08-25 11:53:17 +0400 )редактировать

что то мне подсказывает что дебуг addpac вы не осилите - нанимайте шабашника - быстрее будет

awsswa ( 2014-08-25 12:39:59 +0400 )редактировать

awsswa Где найти такого шабашника

mivan ( 2014-08-27 07:53:52 +0400 )редактировать

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Задан: 2014-08-21 07:48:28 +0400

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Обновлен: Aug 25 '14

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
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