[cisco-voip] h323 trunk between cisco and asterisk
s m
sam.gh1986 at gmail.com
Thu May 7 00:22:14 EDT 2015
hello everybody,
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?
On Thu, Apr 30, 2015 at 1:08 PM, <mtarpey1 at optimum.net> wrote:
> how do u know we r all guys? grow up sexist/rasisixt..............
>
> ----- Original Message -----
> From: s m
> Date: Thursday, April 30, 2015 3:28 am
> Subject: Re: [cisco-voip] h323 trunk between cisco and asterisk
> To: Brian Meade
> Cc: "cisco-voip at puck.nether.net"
>
> > 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
> > ProtocolCallId=A1346065-EE3F11E4-803CFE4D-A6FFC021 at 192.168.0.139
> > 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 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 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
> > 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 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
> > >>>> cisco-voip at puck.nether.net
> > >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>>>
> > >>>>
> > >>>
> > >>
> > >
> >
>
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