[Asterisk-Dev] DTMF and Transfer Call Problem
Sergio Serrano Revuelto
sergio.serrano at avanzada7.com
Mon May 5 08:17:23 MST 2003
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
>
>
>
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-------------- next part --------------
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$192.168.0.155:1052
-- Executing Wait("H323:21089", "1") in new stack
-- Executing Dial("H323:21089", "OH323/8|20|m|t|T") in new stack
WrapH323EndPoint::MakeCall: Making call to 8 using gatekeeper.
WrapH323Connection::WrapH323Connection: Outgoing call with capability G.711-ALaw-64k{hw}
WrapH323Connection::WrapH323Connection: WrapH323Connection created.
WrapH323EndPoint::MakeCall: 5 is calling host 8
WrapH323EndPoint::MakeCall: Call token is ip$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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