<div dir="ltr"><div>hello everybody,<br><br></div>i solve my problem:) it was codec compatibility problem. but it is so strange; if i set
codec g711alaw in cisco router and asterisk, i have the mentioned
problem but if i set codec to transparent in cisco router, every thing
will be ok. is there any difference between g711 codecs which cisco and
asterisk utilize? dose anyone know anything about it?<br><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Apr 30, 2015 at 1:08 PM, <span dir="ltr"><<a href="mailto:mtarpey1@optimum.net" target="_blank">mtarpey1@optimum.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">how do u know we r all guys? grow up sexist/rasisixt..............<span><br><br>----- Original Message -----<br>From: s m <u></u><br></span><span>Date: Thursday, April 30, 2015 3:28 am<br>Subject: Re: [cisco-voip] h323 trunk between cisco and asterisk<br>To: Brian Meade <u></u><br>Cc: "<a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>" <u></u><br><br></span><div><div>> hello guys and thank you for your replies,<br>> <br>> 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></div></div>> *SessionTarget=*<div><div><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>> as you see, SessionTarge feild for h323 leg is empty. i think it <br>> is not<br>> normal, is it? how should i fix it?<br>> i do not have "no ip address trusted authenticate" command in <br>> voice service<br>> voip.<br>> <br>> thanks for your attention.<br>> SAM<br>> <br></div></div><span>> On Wed, Apr 29, 2015 at 7:33 PM, Brian Meade <u></u> wrote:<br>> <br>> > "network 'C0A80047'H" is the IP address. It's just in hex. <br>> That would be<br>> > 192.168.0.71.<br>> ><br>> > Can you send the full H.245 exchange for a call? That should <br>> show us<br>> > where it is failing. We'll want to make sure it gets all the <br>> way yo both<br>> > sides sending OpenLogicalChannelAcks.<br>> ><br></span><div><div>> > On Wed, Apr 29, 2015 at 1:14 AM, s m <u></u> wrote:<br>> ><br>> >> thank you Brian, yes i have set bind address. when i enable h245<br>> >> 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::=<br>> >> 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>> >> i think it is problem. cisco does not know where should send <br>> rtp packets.<br>> >> am i right??? do you have any hint about it???<br>> >><br>> >> thank you for your attention.<br>> >> SAM<br>> >><br>> >> On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade <br></div></div><span>> <u></u> wrote:<br>> >><br>> >>> Do you have "h323-gateway voip bind srcaddr x.x.x.x" <br>> configured on an<br>> >>> interface?<br>> >>><br>> >>> You'll want to run "debug h245 asn1" to see if media <br>> negotiations as<br>> >>> well.<br>> >>><br></span><div><div>> >>> On Tue, Apr 28, 2015 at 3:55 AM, s m <u></u> wrote:<br>> >>><br>> >>>> hello guys,<br>> >>>><br>> >>>> i want to have h323 trunk between cisco 2800 and asterisk <br>> 11.13.1 with<br>> >>>> ooh323 module. i configured both side and have successful <br>> call from cisco<br>> >>>> to asterisk. but when call comes from asterisk to cisco, my <br>> phone rings but<br>> >>>> no audio is heard and call is disconnected after 5 second. <br>> i enable "debug<br>> >>>> voice rtp" in cisco and see the source address for <br>> receiving rtp packets is<br>> >>>> 0.0.0.0<br>> >>>><br>> >>>> Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0),<br>> >>>> d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9<br>> >>>><br>> >>>> any body knows how should i fix it?<br>> >>>><br>> >>>> 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>> >>>> any comments or hints are really appreciated.<br>> >>>> SAM<br>> >>>><br>> >>>><br>> >>>> _______________________________________________<br>> >>>> cisco-voip mailing list<br>> >>>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>> >>>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>> >>>><br>> >>>><br>> >>><br>> >><br>> ><br>> <u></u><u></u><u></u><u></u><u></u><u></u><u></u>
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