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Vg224 + asterisk нет входящих call rejected от cisco

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=forcefport,comedia [multifon-out] dtmfmode=inband username=792xxxxxxxx type=peer secret=NvnZrzRO host=sbc.megafon.ru fromuser=792xxxxxxxx fromdomain = multifon.ru port=5060 nat=nat=forcefport,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(companytree,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@companytree: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 == 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 (PRECEDENCELEVELNONE)

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 discpioff 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 discpioff 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

Vg224 + asterisk нет входящих call rejected от cisco

[general]
tcpenable=yes
allow=alaw
register => tcp://792xxxxxxxx@multifon.ru:NvnZrzRO:792xxxxxxxx@sbc.megafon.ru:5060/792xxxxxxxx
nat=forcefport,comedia
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=forcefport,comedia
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;redirectmedia=no

extensions.conf

extensions.conf
[general] [multifon-in] exten=> 792xxxxxxxx,1,Goto(companytree,s,1) 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/${EXTEN}) exten=>_7XXXXXXXXXX,1,Dial(SIP/multifon-out/${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/${EXTEN},60,rt) exten=>_1XXX,1,Dial(SIP/${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

asterisk

-- Executing [s@companytree:3] [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 Unmonitored

лог cisco ~~vg224~~ vg224

GENERIC: SetupTime=01:44:32.623 SetupTime=*01:44:32.623 UTC Thu Mar 4 1993 Index=19 PeerAddress=79* PeerAddress=79********* PeerSubAddress= PeerId=100 PeerIfIndex=31 LogicalIfIndex=0 DisconnectCause=15 DisconnectText=call rejected (21) ConnectTime=0 DisconnectTime=01:44:32.743 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 FastConnect=FALSE AnnexE=FALSE Separate H245 Connection=FALSE Connection=FALSE H245 Tunneling=FALSE 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** OriginalCallingNumber=79********* OriginalCallingOctet=0x0 OriginalCalledNumber=1001 OriginalCalledOctet=0x0 OriginalRedirectCalledNumber= OriginalRedirectCalledOctet=0x80 TranslatedCallingNumber=79* TranslatedCallingNumber=79********* TranslatedCallingOctet=0x0 TranslatedCalledNumber=1001 TranslatedCalledOctet=0x0 TranslatedRedirectCalledNumber= TranslatedRedirectCalledOctet=0x80 GwReceivedCalledNumber=1001 GwReceivedCalledOctet3=0x0 GwReceivedCallingNumber=79** GwReceivedCallingNumber=79********* GwReceivedCallingOctet3=0x0 GwReceivedCallingOctet3a=0x80 MediaInactiveDetected=no MediaInactiveTimestamp= MediaControlReceived= LongDurationCallDetected=no LongDurationCallTimerStamp= LongDurationCallDuration= Username=79249206728 MlppServiceDomainNW=0 (none) MlppServiceDomainID= PrecedenceLevel=0 (PRECEDENCELEVELNONE) (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 discpioff 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 discpioff 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 ! endend