<div dir="ltr"><div><div>hello guys and thank you for your replies,<br><br></div>this is the output for "show call active voice" command:<br><br><br>R2#show call active voice <br>Telephony call-legs: 0<br>SIP call-legs: 1<br>H323 call-legs: 1<br>Call agent controlled call-legs: 0<br>SCCP call-legs: 0<br>Multicast call-legs: 0<br>Total call-legs: 2<br><br> GENERIC:<br>SetupTime=11153340 ms<br>Index=1<br>PeerAddress=200<br>PeerSubAddress=<br>PeerId=2<br>PeerIfIndex=17<br>LogicalIfIndex=0<br>ConnectTime=0 ms<br>CallDuration=00:00:00 sec<br>CallState=3<br>CallOrigin=2<br>ChargedUnits=0<br>InfoType=speech<br>TransmitPackets=0<br>TransmitBytes=0<br>ReceivePackets=0<br>ReceiveBytes=0<br>VOIP:<br>ConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152]<br>IncomingConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152]<br>CallID=23<br>RemoteIPAddress=192.168.0.71<br>RemoteUDPPort=0<br>RemoteSignallingIPAddress=192.168.0.71<br>RemoteSignallingPort=12031<br>RemoteMediaIPAddress=0.0.0.0<br>RemoteMediaPort=0<br>RoundTripDelay=0 ms<br>SelectedQoS=best-effort<br>tx_DtmfRelay=h245-alphanumeric<br>FastConnect=FALSE<br><br>AnnexE=FALSE<br><br>Separate H245 Connection=FALSE<br><br>H245 Tunneling=TRUE<br><br>SessionProtocol=cisco<br>ProtocolCallId=<br><b>SessionTarget=</b><br>OnTimeRvPlayout=0<br>GapFillWithSilence=0 ms<br>GapFillWithPrediction=0 ms<br>GapFillWithInterpolation=0 ms<br>GapFillWithRedundancy=0 ms<br>HiWaterPlayoutDelay=0 ms<br>LoWaterPlayoutDelay=0 ms<br>TxPakNumber=0 <br>TxSignalPak=0 <br>TxComfortNoisePak=0 <br>TxDuration=0 <br>TxVoiceDuration=0 <br>RxPakNumber=0 <br>RxSignalPak=0 <br>RxDuration=0 <br>TxVoiceDuration=0 <br>VoiceRxDuration=0 <br>RxOutOfSeq=0 <br>RxLatePak=0 <br>RxEarlyPak=0 <br>PlayDelayCurrent=0 <br>PlayDelayMin=0 <br>PlayDelayMax=0 <br>PlayDelayClockOffset=0 <br>PlayDelayJitter=0 ms<br>PlayErrPredictive=0 <br>PlayErrInterpolative=0 <br>PlayErrSilence=0 <br>PlayErrBufferOverFlow=0 <br>PlayErrRetroactive=0 <br>PlayErrTalkspurt=0 <br>OutSignalLevel=0 <br>InSignalLevel=0 <br>LevelTxPowerMean=0 <br>LevelRxPowerMean=0 <br>LevelBgNoise=0 <br>ERLLevel=0 <br>ACOMLevel=0 <br>ErrRxDrop=0 <br>ErrTxDrop=0 <br>ErrTxControl=0 <br>ErrRxControl=0 <br>ReceiveDelay=0 ms<br>LostPackets=0<br>EarlyPackets=0<br>LatePackets=0<br>SRTP = off<br>VAD = enabled<br>CoderTypeRate=g711ulaw<br>CodecBytes=160<br>Media Setting=flow-through<br>CallerName=200<br>CallerIDBlocked=False<br>OriginalCallingNumber=200<br>OriginalCallingOctet=0x1<br>OriginalCalledNumber=100<br>OriginalCalledOctet=0x81<br>OriginalRedirectCalledNumber=<br>OriginalRedirectCalledOctet=0xFF<br>TranslatedCallingNumber=200<br>TranslatedCallingOctet=0x1<br>TranslatedCalledNumber=100<br>TranslatedCalledOctet=0x81<br>TranslatedRedirectCalledNumber=<br>TranslatedRedirectCalledOctet=0xFF<br>GwReceivedCalledNumber=100<br>GwReceivedCalledOctet3=0x81<br>GwReceivedCallingNumber=200<br>GwReceivedCallingOctet3=0x1<br>GwReceivedCallingOctet3a=0x80<br>MediaInactiveDetected=no<br>MediaInactiveTimestamp=<br>MediaControlReceived=<br>Username=<br><br> GENERIC:<br>SetupTime=11153340 ms<br>Index=2<br>PeerAddress=100<br>PeerSubAddress=<br>PeerId=1<br>PeerIfIndex=16<br>LogicalIfIndex=0<br>ConnectTime=0 ms<br>CallDuration=00:00:00 sec<br>CallState=2<br>CallOrigin=1<br>ChargedUnits=0<br>InfoType=speech<br>TransmitPackets=0<br>TransmitBytes=0<br>ReceivePackets=0<br>ReceiveBytes=0<br>VOIP:<br>ConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152]<br>IncomingConnectionId[0x43444546 0x4748494A 0x4B4C4D4E 0x4F505152]<br>CallID=24<br>RemoteIPAddress=192.168.0.78<br>RemoteUDPPort=0<br>RemoteSignallingIPAddress=192.168.0.78<br>RemoteSignallingPort=5060<br>RemoteMediaIPAddress=0.0.0.0<br>RemoteMediaPort=0<br>RoundTripDelay=0 ms<br>SelectedQoS=best-effort<br>tx_DtmfRelay=inband-voice<br>FastConnect=FALSE<br> <br>AnnexE=FALSE<br><br>Separate H245 Connection=FALSE<br><br>H245 Tunneling=FALSE<br><br>SessionProtocol=sipv2<br>ProtocolCallId=<a href="mailto:A1346065-EE3F11E4-803CFE4D-A6FFC021@192.168.0.139" target="_blank">A1346065-EE3F11E4-803CFE4D-A6FFC021@192.168.0.139</a><br>SessionTarget=192.168.0.78<br>OnTimeRvPlayout=0<br>GapFillWithSilence=0 ms<br>GapFillWithPrediction=0 ms<br>GapFillWithInterpolation=0 ms<br>GapFillWithRedundancy=0 ms<br>HiWaterPlayoutDelay=0 ms<br>LoWaterPlayoutDelay=0 ms<br>TxPakNumber=0 <br>TxSignalPak=0 <br>TxComfortNoisePak=0 <br>TxDuration=0 <br>TxVoiceDuration=0 <br>RxPakNumber=0 <br>RxSignalPak=0 <br>RxDuration=0 <br>TxVoiceDuration=0 <br>VoiceRxDuration=0 <br>RxOutOfSeq=0 <br>RxLatePak=0 <br>RxEarlyPak=0 <br>PlayDelayCurrent=0 <br>PlayDelayMin=0 <br>PlayDelayMax=0 <br>PlayDelayClockOffset=0 <br>PlayDelayJitter=0 ms<br>PlayErrPredictive=0 <br>PlayErrInterpolative=0 <br>PlayErrSilence=0 <br>PlayErrBufferOverFlow=0 <br>PlayErrRetroactive=0 <br>PlayErrTalkspurt=0 <br>OutSignalLevel=0 <br>InSignalLevel=0 <br>LevelTxPowerMean=0 <br>LevelRxPowerMean=0 <br>LevelBgNoise=0 <br>ERLLevel=0 <br>ACOMLevel=0 <br>ErrRxDrop=0 <br>ErrTxDrop=0 <br>ErrTxControl=0 <br>ErrRxControl=0 <br>ReceiveDelay=0 ms<br>LostPackets=0<br>EarlyPackets=0<br>LatePackets=0<br>SRTP = off<br>VAD = enabled<br>CoderTypeRate=g711ulaw<br>CodecBytes=160<br>Media Setting=flow-through<br>AlertTimepoint=11153370 ms<br>CallerName=200<br>CallerIDBlocked=False<br>OriginalCallingNumber=200<br>OriginalCallingOctet=0x1<br>OriginalCalledNumber=100<br>OriginalCalledOctet=0x81<br>OriginalRedirectCalledNumber=<br>OriginalRedirectCalledOctet=0xFF<br>TranslatedCallingNumber=200<br>TranslatedCallingOctet=0x1<br>TranslatedCalledNumber=100<br>TranslatedCalledOctet=0x81<br>TranslatedRedirectCalledNumber=<br>TranslatedRedirectCalledOctet=0xFF<br>GwReceivedCalledNumber=100<br>GwReceivedCalledOctet3=0x81<br>GwOutpulsedCalledNumber=100<br>GwOutpulsedCalledOctet3=0x81<br>GwReceivedCallingNumber=200<br>GwReceivedCallingOctet3=0x1<br>GwReceivedCallingOctet3a=0x80<br>GwOutpulsedCallingNumber=200<br>GwOutpulsedCallingOctet3=0x1<br>GwOutpulsedCallingOctet3a=0x80<br>MediaInactiveDetected=no<br>MediaInactiveTimestamp=<br>MediaControlReceived=<br>Username=192.168.0.71<br>Telephony call-legs: 0<br>SIP call-legs: 1<br>H323 call-legs: 1<br>Call agent controlled call-legs: 0<br>SCCP call-legs: 0<br>Multicast call-legs: 0<br>Total call-legs: 2<br><br><br></div><div>as you see, SessionTarge feild for h323 leg is empty. i think it is not normal, is it? how should i fix it? <br></div><div>i do not have "no ip address trusted authenticate" command in voice service voip.<br><br></div><div>thanks for your attention.<br></div><div>SAM<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Apr 29, 2015 at 7:33 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><span style="font-size:12.8000001907349px">"network 'C0A80047'H" is the IP address. It's just in hex. That would be 192.168.0.71.</span><br><div><span style="font-size:12.8000001907349px"><br></span></div><div><span style="font-size:12.8000001907349px">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.</span></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Apr 29, 2015 at 1:14 AM, s m <span dir="ltr"><<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>thank you Brian, yes i have set bind address. when i enable h245 debugging, all messages have no ip address like this:<br>value OpenLogicalChannel ::= <br> {<br> forwardLogicalChannelNumber 1001<br> forwardLogicalChannelParameters <br> {<br> dataType nullData : NULL<br> multiplexParameters none : NULL<br> }<br> reverseLogicalChannelParameters <br> {<br> dataType audioData : g711Ulaw64k : 20<br> multiplexParameters h2250LogicalChannelParameters : <br> {<br> sessionID 1<br> mediaChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17680<br> }<br> mediaControlChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17681<br> }<br> }<br> }<br> }<br><br><br><br>Apr 29 05:09:23.499: H245 FS OLC INCOMING ENCODE BUFFER::= 0003E90C6013800A04000100C0A800474511<br>Apr 29 05:09:23.499: <br>Apr 29 05:09:23.499: H245 FS OLC INCOMING PDU ::=<br><br>value OpenLogicalChannel ::= <br> {<br> forwardLogicalChannelNumber 1002<br> forwardLogicalChannelParameters <br> {<br> dataType audioData : g711Ulaw64k : 20<br> multiplexParameters h2250LogicalChannelParameters : <br> {<br> sessionID 1<br> mediaControlChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17681<br> }<br> }<br> }<br> }<br><br></div>i think it is problem. cisco does not know where should send rtp packets. am i right??? do you have any hint about it???<br><br></div>thank you for your attention.<br></div>SAM<br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Do you have "<span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px">h323-gateway voip bind srcaddr x.x.x.x" configured on an interface?</span><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px"><br></span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px">You'll want to run "debug h245 asn1" to see if media negotiations as well.</span></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Apr 28, 2015 at 3:55 AM, s m <span dir="ltr"><<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div><div><div><div><div>hello guys,<br><br></div>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<br><br> 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<br><br></div>any body knows how should i fix it?<br><br></div>this is my cisco config:<br><br>voice service voip <br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> sip<br>!<br>!<br>!<br>voice class codec 1<br> codec preference 1 g711ulaw<br> codec preference 2 g711alaw<br> codec preference 3 g729r8<br>!<br>dial-peer voice 1 voip<br> destination-pattern 2.+<br> voice-class codec 1<br> session protocol sipv2<br> session target ipv4:192.168.0.240<br>!<br>dial-peer voice 2 voip<br> destination-pattern 1.+<br> voice-class codec 1<br> session target ipv4:<a href="http://192.168.0.71:1720" target="_blank">192.168.0.71:1720</a><br><br></div>any comments or hints are really appreciated.<br></div>SAM<br><div><div><br></div></div></div>
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