[cisco-voip] h323 trunk between cisco and asterisk
s m
sam.gh1986 at gmail.com
Thu Apr 30 03:27:32 EDT 2015
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 <bmeade90 at vt.edu> 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 <sam.gh1986 at gmail.com> 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 <bmeade90 at vt.edu> 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 <sam.gh1986 at gmail.com> 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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