[Asterisk-Dev] DTMF and Transfer Call Problem

[Michael Manousos]

Sergio Serrano Revuelto wrote:
> Thank you very much, 
> 	you have reason. 
> 	With # key transfer works fine but when I use flash key transfer
> works wrong. Something is something

Well, this can't be handled by Asterisk (only # is handled),
although it could be implemented in the channel driver.


Michael.

> 
> 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 19:07
> Para: asterisk-dev at lists.digium.com
> Asunto: Re: [Asterisk-Dev] DTMF and Transfer Call Problem
> 
> 
> 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
>>>>
>>>>


>>
>>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.
> 
> 
>