hello guys and thank you for your replies, this is the output for "show call active voice" command:
R2#show call active voice Telephony call-legs: 0 SIP call-legs: 1 H323 call-legs: 1 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 GENERIC: SetupTime=11153340 ms Index=1 PeerAddress=200 PeerSubAddress= PeerId=2 PeerIfIndex=17 LogicalIfIndex=0 ConnectTime=0 ms CallDuration=00:00:00 sec CallState=3 CallOrigin=2 ChargedUnits=0 InfoType=speech TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=0 VOIP: ConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152] IncomingConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152] CallID=23 RemoteIPAddress=192.168.0.71 RemoteUDPPort=0 RemoteSignallingIPAddress=192.168.0.71 RemoteSignallingPort=12031 RemoteMediaIPAddress=0.0.0.0 RemoteMediaPort=0 RoundTripDelay=0 ms SelectedQoS=best-effort tx_DtmfRelay=h245-alphanumeric FastConnect=FALSE AnnexE=FALSE Separate H245 Connection=FALSE H245 Tunneling=TRUE SessionProtocol=cisco ProtocolCallId= *SessionTarget=* OnTimeRvPlayout=0 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=0 ms LoWaterPlayoutDelay=0 ms TxPakNumber=0 TxSignalPak=0 TxComfortNoisePak=0 TxDuration=0 TxVoiceDuration=0 RxPakNumber=0 RxSignalPak=0 RxDuration=0 TxVoiceDuration=0 VoiceRxDuration=0 RxOutOfSeq=0 RxLatePak=0 RxEarlyPak=0 PlayDelayCurrent=0 PlayDelayMin=0 PlayDelayMax=0 PlayDelayClockOffset=0 PlayDelayJitter=0 ms PlayErrPredictive=0 PlayErrInterpolative=0 PlayErrSilence=0 PlayErrBufferOverFlow=0 PlayErrRetroactive=0 PlayErrTalkspurt=0 OutSignalLevel=0 InSignalLevel=0 LevelTxPowerMean=0 LevelRxPowerMean=0 LevelBgNoise=0 ERLLevel=0 ACOMLevel=0 ErrRxDrop=0 ErrTxDrop=0 ErrTxControl=0 ErrRxControl=0 ReceiveDelay=0 ms LostPackets=0 EarlyPackets=0 LatePackets=0 SRTP = off VAD = enabled CoderTypeRate=g711ulaw CodecBytes=160 Media Setting=flow-through CallerName=200 CallerIDBlocked=False OriginalCallingNumber=200 OriginalCallingOctet=0x1 OriginalCalledNumber=100 OriginalCalledOctet=0x81 OriginalRedirectCalledNumber= OriginalRedirectCalledOctet=0xFF TranslatedCallingNumber=200 TranslatedCallingOctet=0x1 TranslatedCalledNumber=100 TranslatedCalledOctet=0x81 TranslatedRedirectCalledNumber= TranslatedRedirectCalledOctet=0xFF GwReceivedCalledNumber=100 GwReceivedCalledOctet3=0x81 GwReceivedCallingNumber=200 GwReceivedCallingOctet3=0x1 GwReceivedCallingOctet3a=0x80 MediaInactiveDetected=no MediaInactiveTimestamp= MediaControlReceived= Username= GENERIC: SetupTime=11153340 ms Index=2 PeerAddress=100 PeerSubAddress= PeerId=1 PeerIfIndex=16 LogicalIfIndex=0 ConnectTime=0 ms CallDuration=00:00:00 sec CallState=2 CallOrigin=1 ChargedUnits=0 InfoType=speech TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=0 VOIP: ConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152] IncomingConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152] CallID=24 RemoteIPAddress=192.168.0.78 RemoteUDPPort=0 RemoteSignallingIPAddress=192.168.0.78 RemoteSignallingPort=5060 RemoteMediaIPAddress=0.0.0.0 RemoteMediaPort=0 RoundTripDelay=0 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=FALSE AnnexE=FALSE Separate H245 Connection=FALSE H245 Tunneling=FALSE SessionProtocol=sipv2 [email protected] SessionTarget=192.168.0.78 OnTimeRvPlayout=0 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=0 ms LoWaterPlayoutDelay=0 ms TxPakNumber=0 TxSignalPak=0 TxComfortNoisePak=0 TxDuration=0 TxVoiceDuration=0 RxPakNumber=0 RxSignalPak=0 RxDuration=0 TxVoiceDuration=0 VoiceRxDuration=0 RxOutOfSeq=0 RxLatePak=0 RxEarlyPak=0 PlayDelayCurrent=0 PlayDelayMin=0 PlayDelayMax=0 PlayDelayClockOffset=0 PlayDelayJitter=0 ms PlayErrPredictive=0 PlayErrInterpolative=0 PlayErrSilence=0 PlayErrBufferOverFlow=0 PlayErrRetroactive=0 PlayErrTalkspurt=0 OutSignalLevel=0 InSignalLevel=0 LevelTxPowerMean=0 LevelRxPowerMean=0 LevelBgNoise=0 ERLLevel=0 ACOMLevel=0 ErrRxDrop=0 ErrTxDrop=0 ErrTxControl=0 ErrRxControl=0 ReceiveDelay=0 ms LostPackets=0 EarlyPackets=0 LatePackets=0 SRTP = off VAD = enabled CoderTypeRate=g711ulaw CodecBytes=160 Media Setting=flow-through AlertTimepoint=11153370 ms CallerName=200 CallerIDBlocked=False OriginalCallingNumber=200 OriginalCallingOctet=0x1 OriginalCalledNumber=100 OriginalCalledOctet=0x81 OriginalRedirectCalledNumber= OriginalRedirectCalledOctet=0xFF TranslatedCallingNumber=200 TranslatedCallingOctet=0x1 TranslatedCalledNumber=100 TranslatedCalledOctet=0x81 TranslatedRedirectCalledNumber= TranslatedRedirectCalledOctet=0xFF GwReceivedCalledNumber=100 GwReceivedCalledOctet3=0x81 GwOutpulsedCalledNumber=100 GwOutpulsedCalledOctet3=0x81 GwReceivedCallingNumber=200 GwReceivedCallingOctet3=0x1 GwReceivedCallingOctet3a=0x80 GwOutpulsedCallingNumber=200 GwOutpulsedCallingOctet3=0x1 GwOutpulsedCallingOctet3a=0x80 MediaInactiveDetected=no MediaInactiveTimestamp= MediaControlReceived= Username=192.168.0.71 Telephony call-legs: 0 SIP call-legs: 1 H323 call-legs: 1 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 as you see, SessionTarge feild for h323 leg is empty. i think it is not normal, is it? how should i fix it? i do not have "no ip address trusted authenticate" command in voice service voip. thanks for your attention. SAM On Wed, Apr 29, 2015 at 7:33 PM, Brian Meade <[email protected]> wrote: > "network 'C0A80047'H" is the IP address. It's just in hex. That would be > 192.168.0.71. > > Can you send the full H.245 exchange for a call? That should show us > where it is failing. We'll want to make sure it gets all the way yo both > sides sending OpenLogicalChannelAcks. > > On Wed, Apr 29, 2015 at 1:14 AM, s m <[email protected]> wrote: > >> thank you Brian, yes i have set bind address. when i enable h245 >> debugging, all messages have no ip address like this: >> value OpenLogicalChannel ::= >> { >> forwardLogicalChannelNumber 1001 >> forwardLogicalChannelParameters >> { >> dataType nullData : NULL >> multiplexParameters none : NULL >> } >> reverseLogicalChannelParameters >> { >> dataType audioData : g711Ulaw64k : 20 >> multiplexParameters h2250LogicalChannelParameters : >> { >> sessionID 1 >> mediaChannel unicastAddress : iPAddress : >> { >> network 'C0A80047'H >> tsapIdentifier 17680 >> } >> mediaControlChannel unicastAddress : iPAddress : >> { >> network 'C0A80047'H >> tsapIdentifier 17681 >> } >> } >> } >> } >> >> >> >> Apr 29 05:09:23.499: H245 FS OLC INCOMING ENCODE BUFFER::= >> 0003E90C6013800A04000100C0A800474511 >> Apr 29 05:09:23.499: >> Apr 29 05:09:23.499: H245 FS OLC INCOMING PDU ::= >> >> value OpenLogicalChannel ::= >> { >> forwardLogicalChannelNumber 1002 >> forwardLogicalChannelParameters >> { >> dataType audioData : g711Ulaw64k : 20 >> multiplexParameters h2250LogicalChannelParameters : >> { >> sessionID 1 >> mediaControlChannel unicastAddress : iPAddress : >> { >> network 'C0A80047'H >> tsapIdentifier 17681 >> } >> } >> } >> } >> >> i think it is problem. cisco does not know where should send rtp packets. >> am i right??? do you have any hint about it??? >> >> thank you for your attention. >> SAM >> >> On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade <[email protected]> wrote: >> >>> Do you have "h323-gateway voip bind srcaddr x.x.x.x" configured on an >>> interface? >>> >>> You'll want to run "debug h245 asn1" to see if media negotiations as >>> well. >>> >>> On Tue, Apr 28, 2015 at 3:55 AM, s m <[email protected]> wrote: >>> >>>> hello guys, >>>> >>>> i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with >>>> ooh323 module. i configured both side and have successful call from cisco >>>> to asterisk. but when call comes from asterisk to cisco, my phone rings but >>>> no audio is heard and call is disconnected after 5 second. i enable "debug >>>> voice rtp" in cisco and see the source address for receiving rtp packets is >>>> 0.0.0.0 >>>> >>>> Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0), >>>> d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9 >>>> >>>> any body knows how should i fix it? >>>> >>>> this is my cisco config: >>>> >>>> voice service voip >>>> allow-connections h323 to sip >>>> allow-connections sip to h323 >>>> allow-connections sip to sip >>>> sip >>>> ! >>>> ! >>>> ! >>>> voice class codec 1 >>>> codec preference 1 g711ulaw >>>> codec preference 2 g711alaw >>>> codec preference 3 g729r8 >>>> ! >>>> dial-peer voice 1 voip >>>> destination-pattern 2.+ >>>> voice-class codec 1 >>>> session protocol sipv2 >>>> session target ipv4:192.168.0.240 >>>> ! >>>> dial-peer voice 2 voip >>>> destination-pattern 1.+ >>>> voice-class codec 1 >>>> session target ipv4:192.168.0.71:1720 >>>> >>>> any comments or hints are really appreciated. >>>> SAM >>>> >>>> >>>> _______________________________________________ >>>> cisco-voip mailing list >>>> [email protected] >>>> https://puck.nether.net/mailman/listinfo/cisco-voip >>>> >>>> >>> >> >
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