[Asterisk-Users] Transfer (again!)

[WipeOut .]

These are probably more issues for grandstream.. Maybe mail support at grandstream.com with the issues about dropping both calls when the phone is hung up..

Later

> Hello,
> 
> I am building an asterisk PBX with some stuff to make a usable VOIP / 
> PSTN Gateway. I use the following devices:
> SIP Phones from GrandStream for VOIP side
> OpenLine4 from voicetronix for PSTN Side
> 
> I am building things step by step with some priorities.
> 
> I have now a working system able to place and receive calls from/to pstn.
> 
> Before attempting to bring other functions (like voice messaging) up i 
> want to have a proper call transfert functionnality.
> I can't have either blind transfert or consultative transfert working 
> properly.  I am VERY interested in consultative transfert but I don't 
> see where and how 'transfer', 'flash' or 'hold' keys and handle in 
> asterisk code.
> 
> What I would like to do is:
> A and B are taking each other
> A press flash key: B listens music (thet works) and A can call C
> A and C can talk each other but there is no mean for A to transfert B to 
> C. Where should I patch the code to be able to do that?
> Here A can talk either with B or C by pressing on 'Flash' Key but can't 
> hang up any call.
> 
> 
> IF C is Unavalaible, I haven't seen how to get B back
> 
> I welcome any idea about transfert application as it is a main issue for me:
> AGI application,
> Use of Transfert built in
> Proper use of extension.conf file,
> Patch to the source code of asterisk (I am able to do such a patch but I 
> don't know where to look... chan_sip? apps directory, other?)
> 
> Best ragards,
> 
> Daniel
> 
> -- 
> Daniel ANDRE (mailto:dandre at iris-tech.fr)
> IRIS Technologies - http://www.iris-tech.com
> Serveur kwartz - http://www.kwartz.com
> 
> 
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

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