 | 1 | изначальная версия редактировать | |
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 настройки Addpac gs-1002
APOS(tm) configuration saved from vty 2014/08/21 13:27:52
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
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
Voice port configuration.
GSM
voice-port 0/0
input gain 2
output gain 2
connection plar 110
dial-tone-generate
caller-id enable
GSM
voice-port 0/1
input gain 2
output gain 2
connection plar 110
dial-tone-generate
caller-id enable
FXO
voice-port 0/2
input gain 2
output gain 2
connection plar 357091
ring detect-timeout 80
caller-id enable
caller-id type etsi
FXO
voice-port 0/3
input gain 2
output gain 2
connection plar 357093
ring detect-timeout 80
caller-id enable
caller-id type etsi
service port group configuration.
Pots peer configuration.
dial-peer voice 900 pots
destination-pattern 91T
port 0/0
no register e164
translate-outgoing called-number 900
dial-peer voice 901 pots
destination-pattern 92T
port 0/1
no register e164
translate-outgoing called-number 901
dial-peer voice 2024 pots
destination-pattern T
port 0/2
no register e164
preference 7
dial-peer voice 2025 pots
destination-pattern T
port 0/3
no register e164
preference 8
Voip peer configuration.
dial-peer voice 10100 voip
destination-pattern T
session target ip 192.168.211.3 5060
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
description 192.168.211.3
translate-outgoing called-number 10100
gatekeeper
Gateway configuration.
gateway
h323-id voip.192.168.211.32
no ignore-msg-from-other-gk
Codec classes configuration.
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
Translation Rule configuration.
translation-rule 900
rule 0 91T T
translation-rule 10100
rule 0 T T
translation-rule 901
rule 0 92T T
SIP UA configuration.
sip-ua
sip-server 192.168.211.3 5060 126
register e164
Tones
SMS delivery configuration
sms-delivery
line console
line vty
mobile dev-restart-by-unreg 300
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
mobile 0/0
gsm sms-language utf8
mobile 0/1
gsm sms-language utf8
| | 2 | No.2 Revision редактировать |
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 кодек,Через FXO все идет на ура
ура
настройки Addpac gs-1002
APOS(tm) configuration saved from vty 2014/08/21 13:27:52
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
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
Voice port configuration.
GSM
voice-port 0/0
input gain 2
output gain 2
connection plar 110
dial-tone-generate
caller-id enable
GSM
voice-port 0/1
input gain 2
output gain 2
connection plar 110
dial-tone-generate
caller-id enable
FXO
voice-port 0/2
input gain 2
output gain 2
connection plar 357091
ring detect-timeout 80
caller-id enable
caller-id type etsi
FXO
voice-port 0/3
input gain 2
output gain 2
connection plar 357093
ring detect-timeout 80
caller-id enable
caller-id type etsi
service port group configuration.
Pots peer configuration.
dial-peer voice 900 pots
destination-pattern 91T
port 0/0
no register e164
translate-outgoing called-number 900
dial-peer voice 901 pots
destination-pattern 92T
port 0/1
no register e164
translate-outgoing called-number 901
dial-peer voice 2024 pots
destination-pattern T
port 0/2
no register e164
preference 7
dial-peer voice 2025 pots
destination-pattern T
port 0/3
no register e164
preference 8
Voip peer configuration.
dial-peer voice 10100 voip
destination-pattern T
session target ip 192.168.211.3 5060
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
description 192.168.211.3
translate-outgoing called-number 10100
gatekeeper
Gateway configuration.
gateway
h323-id voip.192.168.211.32
no ignore-msg-from-other-gk
Codec classes configuration.
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
Translation Rule configuration.
translation-rule 900
rule 0 91T T
translation-rule 10100
rule 0 T T
translation-rule 901
rule 0 92T T
SIP UA configuration.
sip-ua
sip-server 192.168.211.3 5060 126
register e164
Tones
SMS delivery configuration
sms-delivery
line console
line vty
mobile dev-restart-by-unreg 300
mobile dev-restart-by-unknown-error
mobile cell-monitor 30
mobile 0/0
gsm sms-language utf8
mobile 0/1
gsm sms-language utf8
| | 3 | Добавление дебагов звонка редактировать |
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 ! 2014/08/2113:27:5223:42:35 ! version 8.51.010 ! hostname GS1002 clock timezone Chita 10 ! username root password routeradministratoradministrator username guest password guestuseruser ! ! script ntpdatedefaultdefault resynchronize 100 server ipus.pool.ntp.orgus.pool.ntp.org ! interfaceLoopback0Loopback0 ip address 127.0.0.1255.0.0.0255.0.0.0 ! interfaceFastEthernet0/0FastEthernet0/0 ip address 192.168.211.32255.255.255.0255.255.255.0 speedautoauto noqos-controlqos-control ! interfaceFastEthernet0/1FastEthernet0/1 no ipaddressaddress speedautoauto noqos-controlqos-control ! interfaceFastEthernet0/1:1FastEthernet0/1:1 ip address 192.168.10.1255.255.255.0255.255.255.0 ! ip route 0.0.0.0 0.0.0.0 192.168.211.11010 ! ! ! ! ftpserverserver httpserverserver ! loggingcommandcommand logging event4-warning4-warning loggingonon ! ! ! ! ! VoIP configuration.! ! ! Voice service voip configuration.! voice service voipprotocolsipsip dtmf-relayrfc-2833rfc-2833 fax protocol t38 redundancy 0fax rate 9600h323 call start fasth323 call tunnel enabletimeout tinit 15 timeout tidt 5 static-jitter-buffer 35 ignore-dtmf-abcd-tone no call-barringunconfigured-ip-addressunconfigured-ip-address no voip-inbound-call-barringenableenable ! ! ! Voice port configuration.! ! GSMvoice-port 0/0connection plarinput gain 2
output gain 2
110201 caller-id enabledial-tone-generate
caller-id name disable ! ! ! GSMvoice-port 0/1connection plarinput gain 2
output gain 2
110202 caller-id enabledial-tone-generate
caller-id name disable ! ! ! FXOvoice-port 0/2connection plarinput gain 2
output gain 2
357091203 ring detect-timeout 80caller-id enablecaller-idtype etsiname disable ! ! ! FXOvoice-port 0/3connection plarinput gain 2
output gain 2
357093204 ring detect-timeout 80caller-id enablecaller-idtype etsiname disable ! ! ! ! ! service port group configuration.! ! ! ! Pots peer configuration.! dial-peer voice 900potspots destination-pattern91T01T port 0/0no register e164translate-outgoing called-number 900! dial-peer voice 901potspots destination-pattern92T02T port 0/1no register e164translate-outgoing called-number 901! dial-peer voice2024 pots902 pots destination-patternT03T port 0/2no register e164translate-outgoing called-number 902 ! dial-peer voicepreference 7
2025 pots903 pots destination-patternT04T port 0/3no register e164translate-outgoing called-number 903 ! ! ! ! Voip peer configuration.preference 8
! dial-peer voice101001 voipdestination-patternT201 session target ip 192.168.211.35060
session protocol sipvoice-class codec 0novadvad dtmf-relay rtp-2833no sid ! dial-peer voice 2 voip destination-pattern 202 session target ip 192.168.211.3description
translate-outgoing called-number 10100
gatekeeper
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.! gatewayh323-id voip.192.168.211.32no ignore-msg-from-other-gkshutdown ! ! ! Codec classes configuration.! voice class codec 0codec preference 1 g711alawcodec preference 2g711ulawg711alaw codec preference 3 g729! ! ! ! Translation Rule configuration.! translation-rule 900rule 091T01T T
! translation-rule10100901 rule 0 02T T
! translation-rule 902 rule 1 03T T
! translation-rule901903 rule 092T03T T
! ! ! ! SIP UA configuration.! sip-uasip-server 192.168.211.3 5060 126remote-party-id ! ! ! Tonesregister e164
! ! ! SMS delivery configuration! sms-delivery! ! ! ! ! lineconsoleconsole ! linevtyvty ! mobile dev-restart-by-unreg300300 no mobiledev-restart-by-unknown-errordev-restart-by-unknown-error mobile cell-monitor3030 ! mobile0/00/0 gsm sms-languageutf8utf8 ! mobile0/10/1 gsm sms-languageutf8utf8 !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/fixlocalprefixfixlocalprefix: 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/fixlocalprefixfixlocalprefix: 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
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.
