[asterisk-dev] SIP Phone Conference from Asterisk.

[Pavel Jezek]

maybe some principes of sip conferencing can be taken from new rfc4245

High-Level Requirements for Tightly Coupled SIP Conferencing
ftp://ftp.rfc-editor.org/in-notes/rfc4245.txt

PJ



Kaloyan Kovachev wrote:
> On Tue, 4 Apr 2006 09:41:12 -0400, Patrick Greene wrote
>
>   
>> We ran into this exact scenario about a year ago and wanted to be able to do
>>     
> what you 
>   
>> want to do, but gave up.
>>
>>     
>
>  it is not impossible, i think. I am not C programmer and may be completely
> wrong, but it should be easy to modify the attended transfer feature to
> forward all 3 parties to a predefined (in features.conf or channel variable)
> extension, which will start MeetMe or other conference app. If instead of
> 'ast_bridge_call_thread_launch(tobj);' something similar to G option of Dial
> is made, when the transferer uses disconnect feature code, then instead of
> hangup to forward the channel to the same extension, but diferent priority to
> separate the transferer.
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