[Asterisk-Users] weird call transfer problem

[C F]

On 4/13/05, Anton Krall <akrall-lists at intruder.com.mx> wrote:
> I have the dtmf commented out on sip.conf
> ;dtmfmode=rfc2833
> And the ata have it configured as info
> 
> The weird thing is tht if I am the one making the call, I CAN do transfers,
> I just cant make them if I am the one receiving the call.
> 
> I understand that removing T will forbid the calling user to transfer but as
> far as I know, I should be able to transfer calls myself...

No you shouldn't unless you have t and you are recieving the call, or
you have T and you are making the call.

 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of El Flynn
> Sent: Miércoles, 13 de Abril de 2005 12:58 a.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] weird call transfer problem
> 
> Anton Krall wrote:
> > Guys.
> >
> > I just had a weird problem. I have my Dial cmd configured with mwtWT
> > as parameters however, a call came in thru a zap channel and I
> > answered on a sip phone. I tried using # as configured on my
> > features.conf file to transfer the call but the "transfer" prompt
> > never came in, so I asked the person on the zap channel to do the same
> > and voila, he did get the transfer prompt and entered and extension,
> > but what happended is that I was the one that got transfered! Not him!
> So..... Any ideas whats wrong?
> >
> > The sip phone is an ata, a handytone 286 and zaptel cards.
> >
> > Why cant I do the # transfer and they can but Im the one been transfered?
> >
> 
> The "T" option allows the *calling* user to transfer the call, which is what
> happened to you. The "t" option allows the call recipient to transfer the
> caller to another extension. So to stop that from happening, remove the "T"
> option from the dial command.
> 
> As to why you yourself can't transfer it might have something to do with the
> ATA itself, check what dtmfmode is specified in sip.conf. From the sample
> sip.conf, it says:
> 
> dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
> 
> flynn
> 
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