[Asterisk-Users] weird call transfer problem

[Anton Krall]

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...  

-----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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