[Asterisk-Dev] DTMF and Transfer Call Problem
Michael Manousos
manousos at inaccessnetworks.com
Mon May 5 10:06:42 MST 2003
Sergio Serrano Revuelto wrote:
> My oh323.conf file is the next:
> [general]
> listenAddress=0.0.0.0
> listenPort=1720
> fastStart=no
> h245Tunnelling=no
> h245inSetup=no
> inBandDTMF=no
> silenceSuppression=yes
> jitterMin=20
> jitterMax=100
> ipTos=none
> outboundMax=10
> inboundMax=10
> gatekeeper=192.168.0.204
> userInputMode=STRING
> context=outgoing
> [register]
> alias=nbx1
> gwprefix=9
> gwprefix=7
> [codecs]
> codec=G711A
> frames=20
>
> My gatekeeper.ini is the next:
>
> [Gatekeeper::Main]
> Fourtytwo=42
> TimeToLive=120
> TotalBandwidth=-1
>
> [GkStatus::Auth]
> rule=allow
>
> And the trace is in the attachment. I have probed with flash key and #
> key.
>
>
> Thanks
> srsergio
>
>
>
> -----Mensaje original-----
> De: asterisk-dev-admin at lists.digium.com
> [mailto:asterisk-dev-admin at lists.digium.com] En nombre de Michael
> Manousos
> Enviado el: lunes, 05 de mayo de 2003 16:45
> Para: asterisk-dev at lists.digium.com
> Asunto: Re: [Asterisk-Dev] DTMF and Transfer Call Problem
>
>
> Sergio Serrano Revuelto wrote:
>
>>I'm sorry, transfer fail. I try with gnugk in non-routed mode and also
>
>
>>transfer fails.
>>
>
>
> Have you tried to disable all H.323 v2,v4 features in the channel driver
> (fastStart, H245inSetup, H245Tunneling)?
>
> This feature works just fine, at least with DTMF as H.245 string or
> tone. Can you try it with this DTMF mode?
>
> If the transfer still fails, you could send a log of the Asterisk output
> to take a look.
>
> Michael.
>
>
>
>>-----Mensaje original-----
>>De: asterisk-dev-admin at lists.digium.com
>>[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Michael
>>Manousos Enviado el: lunes, 05 de mayo de 2003 15:45
>>Para: asterisk-dev at lists.digium.com
>>Asunto: Re: [Asterisk-Dev] DTMF and Transfer Call Problem
>>
>>
>>Sergio Serrano Revuelto wrote:
>>
>>
>>>Hi,
>>> I have two problem. First, * doesn't recognize flash key, I need
>>
>>
>>>press flash key and other key to transfer a call. My phones are
>>>configured with RFC2833 signalling, GNUGK is configured in routed mode
>>
>>
>>>and H.245 routed mode, and chan_oh323 is configured without FastStart
>>>and with h.245tunnelling, and RFC2833 signalling.
>>>
>>>The other problem is if I press flash key and other key, then I do a
>>>call transfer, but it occurs the next:
>>> Phone 1 calls to Phone 2, I pickup Phone 2, then I transfer call
>>
>>to
>>
>>
>>>Phone 3 and hangup Phone 2 -> communication crashes
>>> Phone 1 calls to Phone 2, I pickup Phone 2, then I transfer call
>>
>>to
>>
>>
>>>Phone 3 and pickup Phone 3 adn then hangup Phone 2 -> communication
>>>crashes.
>>>
>>
>>
>>What do you mean by saying "communication crashes"? Does Asterisk
>>crash, or just the transfer fail?
>>
>>Something else. Have you try it with the gnugk in non-routed mode?
>>
>>Michael.
>>
>>
>>
>>
>>> I'm desperate, I don't know if it is a bad configuration for *
>>
>>or
>>
>>
>>>gnugk, or ....
>>>
>>>
>>>If anyone could help, I'll be pleasant with him.
>>>
>>>
>>>Thanks in advance
>>>srsergio
>>>
>>>_______________________________________________
>>>Asterisk-Dev mailing list
>>>Asterisk-Dev at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
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>
>
> ------------------------------------------------------------------------
>
> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
> WrapH323Connection::OnReceivedSignalSetup: Received SETUP message...
> WrapH323Connection::OnAnswerCall: User Sergio (5, 5) [192.168.0.155] is calling us...
> WrapH323Connection::OnAnswerCall: Call reference: 21089
> WrapH323Connection::OnAnswerCall: Call token: ip$192.168.0.155:1052/21089
> WrapH323Connection::OnAnswerCall: Call source alias: Sergio (5, 5) [192.168.0.155](29)
> WrapH323Connection::OnAnswerCall: Call dest alias: 708 708(7)
> WrapH323Connection::OnAnswerCall: Call source e164: 5(1)
> WrapH323Connection::OnAnswerCall: Call dest e164: 708(3)
> WrapH323Connection::OnAnswerCall: Remote Party number: 5
> WrapH323Connection::OnAnswerCall: Remote Party name: Sergio (5, 5) [192.168.0.155]
> WrapH323Connection::OnAnswerCall: Remote Party address: 5 at ip$192.168.0.155:1052
> -- Executing Wait("H323:21089", "1") in new stack
> -- Executing Dial("H323:21089", "OH323/8|20|m|t|T") in new stack
Your extensions.conf is wrong. Your Dial, in extensions.conf should
look like that:
Dial,OH323/8|20|mtT
or even better:
Dial(OH323/8,20,mtT)
Also 'T' is not yet implemented in Asterisk.
Michael.
> WrapH323EndPoint::MakeCall: Making call to 8 using gatekeeper.
> WrapH323Connection::WrapH323Connection: Outgoing call with capability G.711-ALaw-64k{hw}
> WrapH323Connection::WrapH323Connection: WrapH323Connection created.
> WrapH323EndPoint::MakeCall: 5 is calling host 8
> WrapH323EndPoint::MakeCall: Call token is ip$localhost/6767
> WrapH323EndPoint::MakeCall: Call reference is 6767
> WrapH323Connection::OnSendSignalSetup: Sending SETUP message...
> WrapH323Connection::OnAlerting: Ringing phone for "192.168.0.158" ...
> -- Called 8
> WrapH323EndPoint::OpenAudioChannel: Direction => RECODER, Buffer => 320
> WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8
> WrapH323EndPoint::OpenAudioChannel: Frames 20
> WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k
> WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:in1(fd=46)
> WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:in1
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskSoundChannel::Open: os_handle 46, mediaFormat 8, frameTime 1
> WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:in1" for recording using 1x160 byte buffers.
> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [6767] : sending G.711-ALaw-64k{hw}
> WrapH323EndPoint::OnStartLogicalChannel: TxFrames = 20
> WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 1
> WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 480
> WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8
> WrapH323EndPoint::OpenAudioChannel: Frame 1
> WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k
> WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:out1(fd=44)
> WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:out1
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskSoundChannel::Open: os_handle 44, mediaFormat 8, frameTime 1
> WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:out1" for playing using 1x8 byte buffers.
> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [6767] : receiving G.711-ALaw-64k{hw}
> WrapH323EndPoint::OnStartLogicalChannel: RxFrames = 30
> WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 2
> -- H323:6767 answered H323:21089
> WrapH323EndPoint::AnswerCall: Request to answer call with token ip$192.168.0.155:1052/21089
> WrapH323EndPoint::AnswerCall: Call with token ip$192.168.0.155:1052/21089 answered
> WrapH323EndPoint::OpenAudioChannel: Direction => RECODER, Buffer => 320
> WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8
> WrapH323EndPoint::OpenAudioChannel: Frames 20
> WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k
> WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:in0(fd=42)
> WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:in0
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskSoundChannel::Open: os_handle 42, mediaFormat 8, frameTime 1
> WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:in0" for recording using 1x160 byte buffers.
> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [21089] : sending G.711-ALaw-64k{hw}
> WrapH323EndPoint::OnStartLogicalChannel: TxFrames = 20
> WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 3
> WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 480
> WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8
> WrapH323EndPoint::OpenAudioChannel: Frame 1
> WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-ALaw-64k
> WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:out0(fd=40)
> WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:out0
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
> PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
> PAsteriskSoundChannel::Open: os_handle 40, mediaFormat 8, frameTime 1
> WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:out0" for playing using 1x8 byte buffers.
> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [21089] : receiving G.711-ALaw-64k{hw}
> WrapH323EndPoint::OnStartLogicalChannel: RxFrames = 30
> WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 4
> WrapH323EndPoint::OnUserInputString: Received user input string (!) from remote
> WrapH323EndPoint::SendUserInput: Sent user input string (!) using mode 2
> WrapH323EndPoint::OnUserInputString: Received user input string (!) from remote
> WrapH323EndPoint::SendUserInput: Sent user input string (!) using mode 2
> WrapH323EndPoint::OnUserInputString: Received user input string (!) from remote
> WrapH323EndPoint::SendUserInput: Sent user input string (!) using mode 2
> WrapH323EndPoint::OnUserInputString: Received user input string (#) from remote
> WrapH323EndPoint::SendUserInput: Sent user input string (#) using mode 2
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