[Asterisk-Users] weird call transfer problem

El Flynn
Tue, 12 Apr 2005 22:57:37 -0000

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