[prev in list] [next in list] [prev in thread] [next in thread]
List: cisco-voip
Subject: Re: [cisco-voip] h323 trunk between cisco and asterisk
From: s m <sam.gh1986 () gmail ! com>
Date: 2015-04-30 7:39:32
Message-ID: CAA_1SgH+RwmiErLE2y4mKEe95s1pD3okLBrLEz-75PoKcdjHhQ () mail ! gmail ! com
[Download RAW message or body]
[Attachment #2 (multipart/alternative)]
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@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@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@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@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@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@puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>
>>
>
[Attachment #5 (text/html)]
<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>TransmitPac \
kets=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>OriginalCallingNumb \
er=200<br>OriginalCallingOctet=0x1<br>OriginalCalledNumber=100<br>OriginalCalledOctet= \
0x81<br>OriginalRedirectCalledNumber=<br>OriginalRedirectCalledOctet=0xFF<br>Translate \
dCallingNumber=200<br>TranslatedCallingOctet=0x1<br>TranslatedCalledNumber=100<br>Tran \
slatedCalledOctet=0x81<br>TranslatedRedirectCalledNumber=<br>TranslatedRedirectCalledO \
ctet=0xFF<br>GwReceivedCalledNumber=100<br>GwReceivedCalledOctet3=0x81<br>GwReceivedCa \
llingNumber=200<br>GwReceivedCallingOctet3=0x1<br>GwReceivedCallingOctet3a=0x80<br>Med \
iaInactiveDetected=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>TransmitPac \
kets=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>Original \
CallingOctet=0x1<br>OriginalCalledNumber=100<br>OriginalCalledOctet=0x81<br>OriginalRe \
directCalledNumber=<br>OriginalRedirectCalledOctet=0xFF<br>TranslatedCallingNumber=200 \
<br>TranslatedCallingOctet=0x1<br>TranslatedCalledNumber=100<br>TranslatedCalledOctet= \
0x81<br>TranslatedRedirectCalledNumber=<br>TranslatedRedirectCalledOctet=0xFF<br>GwRec \
eivedCalledNumber=100<br>GwReceivedCalledOctet3=0x81<br>GwOutpulsedCalledNumber=100<br \
>GwOutpulsedCalledOctet3=0x81<br>GwReceivedCallingNumber=200<br>GwReceivedCallingOctet \
> 3=0x1<br>GwReceivedCallingOctet3a=0x80<br>GwOutpulsedCallingNumber=200<br>GwOutpulse \
> dCallingOctet3=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>
<br></div></div>_______________________________________________<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></blockquote></div><br></div> </blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>
_______________________________________________
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
[prev in list] [next in list] [prev in thread] [next in thread]
Configure |
About |
News |
Add a list |
Sponsored by KoreLogic