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

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