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спросил 2011-11-05 10:27:11 +0400

annadiz85 Gravatar annadiz85

Elastix + addpac GS1001 не проходят входящие

Добрый день! пытаюсь подружить addpac gs1001, исходящие проходят а вот входящие почему то нет

конфиг addpac

! ! APOS(tm) configuration saved from vty ! 2011/11/04 15:57:10 ! version 8.51.002 ! hostname GS1001 ! username root password router administrator username guest password guest user ! ! interface Loopback0 ip address 127.0.0.1 255.0.0.0 ! interface FastEthernet0/0 ip address 192.168.0.160 255.255.255.0 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 0.0.0.0 192.168.0.1 10 ! access-list 100 permit ip 192.168.10.0 0.0.0.255 any ! ! ip nat inside source list 100 interface FastEthernet0/0 overload ! ! ! ftp server http server ! dns name-server 192.168.0.1 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 call-barring allow-ip 192.168.0.150 voip-inbound-call-barring enable voip-inbound-call-barring allow-digits 11 11 ! ! ! Voice port configuration. ! ! GSM voice-port 0/0 connection plar 480707 caller-id enable ! ! ! ! ! ! ! ! service port group configuration. ! ! ! ! Pots peer configuration. ! dial-peer voice 1 pots destination-pattern T ! dial-peer voice 900 pots destination-pattern 8T port 0/0 no register e164 ! ! ! ! Voip peer configuration. ! dial-peer voice 10100 voip destination-pattern [0-79]T session target sip-server session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 ! ! ! ! ! ! gatekeeper ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.0.150 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711alaw codec preference 2 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-server 192.168.0.150 hook-flash-info-ignore ! ! ! Tones ! ! ! ! line console ! line vty ! gsm dev-restart-by-unreg 300 ! gsm 0/0 sms-language utf8 ! end

конфиг elastix

trunk 480707 type=friend call-limit=1 host=192.168.0.160 port=5060 fromdomain=192.168.0.160 nat=no qualify=yes canreinvite=no insecure=port,invite dtmfmode=auto dial=SIP/101 context=from-pstn disallow=all allow=alaw&ulaw

контекст user type=friend context=from-pstn host=192.168.0.160 dtmfmode=auto

лог addpac

GS1001# 1 <cep 000000=""> : Call Received 2 <cep 000000=""> : Call Received 3 <cep 000000=""> : Call Initiated : calledNumber() crv(0) total(0) 4 <call 2=""> : * Call Created status(InitiatedByGSM) ver(8.28:2 006-02-06-00-00) time(1320499139) *** 5 <cep 000000=""> : Decode CID : FFFFFF80 E 10 C 2B 37 39 35 31 34 32 30 36 30 36 31 6 <cep 000000=""> : GSM CID : time() callingNumber(79514206061) callingName() 7 <cep 000000=""> : Calling number(79514206061) 8 <cep 000000=""> : Call id(c337b54e-03a1-6eec-8003-0002a40847c4) callNum(2) 9 <call 2=""> : MatchAllProcess After Sorted <0> id(10100) dest([0-79]T) prefer(0) selected(1) <1> id(1) dest(T) prefer(0) selected(0) 10 <call 2=""> : Terminated from(fffffff7) this(Local:ResourceUnavailable) before((null)) forced(0) time(1320499139) 11 <cep 000000=""> : DisconnectCall at Busy 12 <cep 000000=""> : StopSignal 13 <cep 000000=""> : Disconnect (0) 14 <call 2=""> : Terminated from(fffffff7) this(Local:Unknown) before(Local:ResourceUnavailable) forced(0) time(1320499139) 15 <cep 000000=""> : Disconnected(16) at Disconnecting

Elastix + addpac GS1001 не проходят входящие

Добрый день! пытаюсь подружить addpac gs1001, исходящие проходят а вот входящие почему то нет

конфиг addpac

! ! APOS(tm) configuration saved from vty ! 2011/11/04 15:57:10 ! version 8.51.002 ! hostname GS1001 ! username root password router administrator username guest password guest user ! ! interface Loopback0 ip address 127.0.0.1 255.0.0.0 ! interface FastEthernet0/0 ip address 192.168.0.160 255.255.255.0 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 0.0.0.0 192.168.0.1 10 ! access-list 100 permit ip 192.168.10.0 0.0.0.255 any ! ! ip nat inside source list 100 interface FastEthernet0/0 overload ! ! ! ftp server http server ! dns name-server 192.168.0.1 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 call-barring allow-ip 192.168.0.150 voip-inbound-call-barring enable voip-inbound-call-barring allow-digits 11 11 ! ! ! Voice port configuration. ! ! GSM voice-port 0/0 connection plar 480707 caller-id enable ! ! ! ! ! ! ! ! service port group configuration. ! ! ! ! Pots peer configuration. ! dial-peer voice 1 pots destination-pattern T ! dial-peer voice 900 pots destination-pattern 8T port 0/0 no register e164 ! ! ! ! Voip peer configuration. ! dial-peer voice 10100 voip destination-pattern [0-79]T session target sip-server session protocol sip voice-class codec 0 no vad dtmf-relay rtp-2833 ! ! ! ! ! ! gatekeeper ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.0.150 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711alaw codec preference 2 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-server 192.168.0.150 hook-flash-info-ignore ! ! ! Tones ! ! ! ! line console ! line vty ! gsm dev-restart-by-unreg 300 ! gsm 0/0 sms-language utf8 ! end

конфиг elastix

trunk 480707 type=friend call-limit=1 host=192.168.0.160 port=5060 fromdomain=192.168.0.160 nat=no qualify=yes canreinvite=no insecure=port,invite dtmfmode=auto dial=SIP/101 context=from-pstn disallow=all allow=alaw&ulaw

контекст user type=friend context=from-pstn host=192.168.0.160 dtmfmode=auto

лог addpac

GS1001# 1 <cep 000000=""> : Call Received 2 <cep 000000=""> : Call Received 3 <cep 000000=""> : Call Initiated : calledNumber() crv(0) total(0) 4 <call 2=""> : * Call Created status(InitiatedByGSM) ver(8.28:2 006-02-06-00-00) time(1320499139) *** 5 <cep 000000=""> : Decode CID : FFFFFF80 E 10 C 2B 37 39 35 31 34 32 30 36 30 36 31 6 <cep 000000=""> : GSM CID : time() callingNumber(79514206061) callingName() 7 <cep 000000=""> : Calling number(79514206061) 8 <cep 000000=""> : Call id(c337b54e-03a1-6eec-8003-0002a40847c4) callNum(2) 9 <call 2=""> : MatchAllProcess After Sorted <0> id(10100) dest([0-79]T) prefer(0) selected(1) <1> id(1) dest(T) prefer(0) selected(0) 10 <call 2=""> : Terminated from(fffffff7) this(Local:ResourceUnavailable) before((null)) forced(0) time(1320499139) 11 <cep 000000=""> : DisconnectCall at Busy 12 <cep 000000=""> : StopSignal 13 <cep 000000=""> : Disconnect (0) 14 <call 2=""> : Terminated from(fffffff7) this(Local:Unknown) before(Local:ResourceUnavailable) forced(0) time(1320499139) 15 <cep 000000=""> : Disconnected(16) at Disconnecting

Elastix + addpac GS1001 не проходят входящие

Добрый день! день!
пытаюсь подружить addpac gs1001, исходящие проходят а вот входящие почему то нет



конфиг addpac

!
! APOS(tm) configuration saved from vty
!  2011/11/04 15:57:10 
!
version 8.51.002
!
hostname GS1001
!
username root password router administrator
username guest password guest user
!
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.0.160 255.255.255.0
 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 0.0.0.0 192.168.0.1 10
!
access-list 100 permit ip 192.168.10.0 0.0.0.255 any
!
!
ip nat inside source list 100 interface FastEthernet0/0  overload
!
!
!
ftp server
http server
!
dns name-server 192.168.0.1
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
  call-barring allow-ip 192.168.0.150
 voip-inbound-call-barring enable
 voip-inbound-call-barring allow-digits 11 11
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
 connection plar 480707
 caller-id enable
!
! service port group configuration.
!
! Pots peer configuration.
!
dial-peer voice 1 pots
 destination-pattern T
!
dial-peer voice 900 pots
 destination-pattern 8T
 port 0/0
 no register e164
!
!
!
! Voip peer configuration.
!
dial-peer voice 10100 voip
 destination-pattern [0-79]T
 session target sip-server
 session protocol sip
 voice-class codec 0
 no vad
 dtmf-relay rtp-2833
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.0.150
 no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
!
! SIP UA configuration.
!
sip-ua
sip-server 192.168.0.150
 hook-flash-info-ignore
!
!
! Tones
!
line console
!
line vty
!
gsm dev-restart-by-unreg 300
!
gsm 0/0
 sms-language utf8
!
end

end

конфиг elastix

trunk 480707
type=friend
call-limit=1
host=192.168.0.160
port=5060
fromdomain=192.168.0.160
nat=no
qualify=yes
canreinvite=no
insecure=port,invite
dtmfmode=auto
dial=SIP/101
context=from-pstn
disallow=all
allow=alaw&ulaw

allow=alaw&ulaw

контекст user user

type=friend
context=from-pstn
host=192.168.0.160
dtmfmode=auto

dtmfmode=auto

лог addpac

GS1001# 1       <cep 000000=""> <CEP    000000> : Call Received
2       <cep 000000=""> <CEP    000000> : Call Received
3       <cep 000000=""> <CEP    000000> : Call Initiated : calledNumber() crv(0) total(0)
4       <call 2="">      : * <Call   2>      : ******  Call Created status(InitiatedByGSM) ver(8.28:2
006-02-06-00-00) time(1320499139) ***
****
5       <cep 000000=""> <CEP    000000> : Decode CID : FFFFFF80  E 10  C 2B 37 39 35 31 34 32 30 36 30 36 31
6       <cep 000000=""> <CEP    000000> : GSM CID : time() callingNumber(79514206061) callingName()
7       <cep 000000=""> <CEP    000000> : Calling number(79514206061)
8       <cep 000000=""> <CEP    000000> : Call id(c337b54e-03a1-6eec-8003-0002a40847c4) callNum(2)
9       <call 2=""> <Call   2>      : MatchAllProcess After Sorted
                          <0>  id(10100) dest([0-79]T) prefer(0) selected(1)
                          <1>  id(1) dest(T) prefer(0) selected(0)
10      <call 2=""> <Call   2>      : Terminated from(fffffff7) this(Local:ResourceUnavailable) before((null)) forced(0) time(1320499139)
11      <cep 000000=""> <CEP    000000> : DisconnectCall at Busy
12      <cep 000000=""> <CEP    000000> : StopSignal
13      <cep 000000=""> <CEP    000000> : Disconnect (0)
14      <call 2=""> <Call   2>      : Terminated from(fffffff7) this(Local:Unknown) before(Local:ResourceUnavailable) forced(0) time(1320499139)
15      <cep 000000=""> <CEP    000000> : Disconnected(16) at Disconnecting

Disconnecting

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.