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Vg224 + asterisk нет входящих call rejected от cisco

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Здравствуйте! Помогите разобраться! asterisk + мегафон мультифон, конфигурация проверена. Соединено с vg224, два аналоговых телефона. Исходящие с этих телефонов 2/0 и 2/1 через астериск работают, друг другу тоже звонят. sip show peers телефоны тоже показывает. При входящем звонке циско дает call rejected, Everyone is busy/congested/ sip.conf

[general]
tcpenable=yes
allow=alaw
register => tcp://792xxxxxxxx@multifon.ru:NvnZrzRO:792xxxxxxxx@sbc.megafon.ru:5060/792xxxxxxxx
nat=force_fport,comedia


[multifon-out]
dtmfmode=inband
username=792xxxxxxxx
type=peer
secret=NvnZrzRO
host=sbc.megafon.ru
fromuser=792xxxxxxxx
fromdomain = multifon.ru
port=5060
nat=nat=force_fport,comedia
context=multifon-in
insecure=port,invite
insecure=port,invite
canreinvite=no
[1003]
type=friend
host=dynamic
secret=pass1003
context=phones
username=1003
nat=no
allow=alaw
[multifon-in]
type=peer
host=sbc.megafon.ru
dtmfmode=inband
[1001]
type=peer
host=dynamic
context=phones
username=1001
secret=pass1001
[1002]
type=friend
host=dynamic
context=phones
username=1002
secret=pass1002
[vg224]
type=friend
host=dynamic
username=vg224
secret=passvg224
context=phones
;qualify=200
;redirectmedia=no

extensions.conf

[general]
[multifon-in]
exten=> 792xxxxxxxx,1,Goto(company_tree,s,1)
exten=> 792xxxxxxxx,2,Dial(SIP/1001,60,t)
[out]
exten=>_7XXXXXXXXXX,1,Dial(SIP/multifon-out/${EXTEN})
exten=>_8XXXXXXXXXX,1,Dial(SIP/multifon-out/${EXTEN}) 
exten=>_+7XXXXXXXXXX,1,Dial(SIP/multifon-out/${EXTEN})
[local]
exten=>_1XXX,1,Dial(SIP/${EXTEN},60,rt)
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN},60,rt)
exten=>_8XXXXXXXXXX,1,Dial(SIP/${EXTEN},60,rt)
[phones]
include => multifon-in ;
include => local
include => out

asterisk

   -- Executing [s@company_tree:3] Dial("SIP/multifon-out-00000000", "SIP/1001,10,m") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1001
    -- Started music on hold, class 'default', on SIP/multifon-out-00000000
[Mar  6 11:25:13] WARNING[15153][C-00000000]: chan_sip.c:22991 handle_response_invite: Received response: "Forbidden" from '<sip:79249206728@192.168.1.8>;tag=as1b192ad0'
  == Everyone is busy/congested at this time (1:0/0/1)
*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description                      
1001/1001                 192.168.1.7                              D                 5060     Unmonitored                                  
1002/1002                 (Unspecified)                            D                 0        Unmonitored                                  
1003/1003                 (Unspecified)                            D                 0        Unmonitored                                  
101/101                   (Unspecified)                            D                 0        Unmonitored                                  
multifon-in               193.201.229.35                                             5060     Unmonitored                                  
multifon-out/792xxxxxxxx  193.201.229.35                                             5060     Unmonitored                                  
vg224/vg224               (Unspecified)                            D                 0        Unmonitored

лог cisco vg224

GENERIC:
SetupTime=*01:44:32.623 UTC Thu Mar 4 1993
Index=19
PeerAddress=79*********
PeerSubAddress=
PeerId=100
PeerIfIndex=31
LogicalIfIndex=0
DisconnectCause=15  
DisconnectText=call rejected (21)
ConnectTime=0
DisconnectTime=*01:44:32.743 UTC Thu Mar 4 1993
CallDuration=00:00:00 sec
CallOrigin=2
ReleaseSource=7
InternalErrorCode=1.1.228.3.31.0
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=0
VOIP:
ConnectionId[0xA1CF3F4E 0x175811CC 0x80589D9E 0xDE346839]
IncomingConnectionId[0xA1CF3F4E 0x175811CC 0x80589D9E 0xDE346839]
CallID=41
CallReferenceId=0
CallServiceType=Unknown
RTP Loopback Call=FALSE
RemoteIPAddress=192.168.1.8
RemoteUDPPort=11252
RemoteSignallingIPAddress=192.168.1.8
RemoteSignallingPort=5060
RemoteMediaIPAddress=192.168.1.8
RemoteMediaPort=11252
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2
ProtocolCallId=2b62545010d8d40b567da94a1f760d40@192.168.1.8:5060
SessionTarget=192.168.1.8
SafEnabled=FALSE
OnTimeRvPlayout=0
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=0 ms
LoWaterPlayoutDelay=0 ms
ReceiveDelay=0 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711alaw
CodecBytes=160
cvVoIPCallHistoryIcpif=0
MediaSetting=flow-through
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=79*********
OriginalCallingOctet=0x0
OriginalCalledNumber=1001
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x80
TranslatedCallingNumber=79*********
TranslatedCallingOctet=0x0
TranslatedCalledNumber=1001
TranslatedCalledOctet=0x0
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0x80
GwReceivedCalledNumber=1001
GwReceivedCalledOctet3=0x0
GwReceivedCallingNumber=79*********
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x80
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=79249206728
MlppServiceDomainNW=0 (none)
MlppServiceDomainID=
PrecedenceLevel=0 (PRECEDENCE_LEVEL_NONE)

CPA Call History Parameters
  CPA Event Status: DISABLE
конфиг vg224
sh run
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
!
voice class codec 1
 codec preference 1 g711alaw
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
!
voice class codec 1
 codec preference 1 g711alaw
interface FastEthernet0/0
 ip address 192.168.1.7 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
voice-port 2/0
 ren 3
 disconnect-ack
 disc_pi_off
 input gain 10
 output attenuation 10
 playout-delay minimum low
 cptone RU
 timing digit 53
station-id number 1001
 caller-id enable
!
voice-port 2/1
 ren 3
 disconnect-ack
 disc_pi_off
 input gain 10
 output attenuation 10
 playout-delay minimum low
 cptone RU
 timing digit 53
 station-id name P-1002
 station-id number 1002
 caller-id enable
!
dial-peer voice 11 pots
 incoming called-number 7T
 authentication username 1001 password 7 (pass1001 hash)
!
dial-peer voice 12 pots
 authentication username 1002 password 7 (pass1002 hash)
 port 2/1
!
dial-peer voice 100 voip
 huntstop
 destination-pattern 7T
 session protocol sipv2
 session target sip-server
 voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
 no vad
 authentication username 1003 password 7 (pass1003 hash)
!
sip-ua 
 authentication username vg224 password 7 (passvg224 hash)
 retry invite 3
 retry response 10
 timers trying 1000
 timers connect 100
 registrar ipv4:192.168.1.8:5060 expires 3600
 sip-server ipv4:192.168.1.8
!
end
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спросил 2017-03-06 06:24:13 +0400

vg224tester Gravatar vg224tester
1

обновил 2017-03-06 11:28:21 +0400

zzuz Gravatar zzuz flag of Russian Federation
6904 2 6 72
http://line24.ru/

Comments

рекомендую запостить на cisco.com.

meral ( 2017-03-06 14:47:00 +0400 )редактировать

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