[asterisk-users] Transfer feature - howto?

[Colin Anderson]

??? I do it with a Zap channel no problem. In my case,

1. Call comes in from PSTN (Zap channel)
2. Call is routed back out a Zap channel using the Dial() command with the
't' option
3. Asterisk is still in the media stream, so it listens for inband DTMF
4. User presses Hash, Asterisk says "Transfer", user dials extension, hangs
up, call is transferred. 

I suspect in your case it may have something to do with your SIP providers
maybe suppressing DTMF (just guessing) or Asterisk somehow stepping out of
the media stream. You want to test for that by setting up an extension that
will echo back the interpreted DTMF signal then calling the extension via
your SIP providers. But with Zap, definitely works no problem. 

-----Original Message-----
From: Mike [mailto:list at virtutel.ca]
Sent: Wednesday, October 04, 2006 2:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Transfer feature - howto?


Ah.  I'd like to know what others think, but if you're right than it's a
lost cause.

I thought Asterisk kept some sort of control over the call.

Mike 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Steve Glaus
> Sent: October 4, 2006 3:24 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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