[asterisk-users] Transfer feature - howto?

[Colin Anderson]

When you Dial() the cell, are you passing the 't' parameter? Also: When the
call hits the cell, is Asterisk still in the media stream? canreinvite=no
should be explicitly specified in the SIP accounts of your providers in
sip.conf. One more thing: Do you know for a fact that inband DTMF is being
procesed by Asterisk when the call hits the cell?



-----Original Message-----
From: Steve Glaus [mailto:techsupport at peachnet.com]
Sent: Wednesday, October 04, 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer feature - howto?


Mike wrote:
> Hi,
>  
> My setup is the following: Voip provider-----------(SIP 
> DID)----------->Asterisk box----(SIP through a termination 
> provider)--------------->multiple cell phones.
>  
> The cell phones each have their extension (201,202,203,204) and I'd 
> like to be able to have them transfer a call to somebody else.  Ex: 
> Prospect calls extension 201, talks to the salesgy, who forwards him 
> to the tech guru somehow.
>  
> My guess is I have to use the transfer feature found in feature.conf.  
> I tried, no success.  What Wiki page do I need to look at to get 
> details on this?
>  
> Mike
> ------------------------------------------------------------------------
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I don't know if this is even possible. I might be totally wrong but once 
this call is on the cell network, how are you gonna communicate with 
asterisk?? From what I understand, while the voice (RTP) traffic still 
travels through asterisk, You have no access to any kind of signalling. 
Please correct me if I'm way off base here, anyone.
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