[Asterisk-Users] Supervised transfer (almost) with GS phone

[Konrad Gorski]

Brian Capouch wrote:

> Stephen R. Besch wrote:
>
>> I have now tested a (previously suggested) method for doing 
>> supervised transfers using the Grandstream SIP phone. It isn't 
>> perfect, but it works and is very functional. Here are the steps:
>>
>
> When I try this, all goes well until, after putting the original 
> caller on hold and then getting a dialtone, I dial another extension, 
> and then get these errors on the CLI:
>
> find_user: Call from user 'btel' rejected due to usage limit of 1
> find_user: Call from user 'btel' rejected due to usage limit of 1
> find_user: Call from user 'btel' rejected due to usage limit of 1
> find_user: Call from user 'btel' rejected due to usage limit of 1
>
> Then the Grandstream gives me a busy, and my orignal caller is a zombie.
>
> What am I doing wrong?
>
> Thx.
>
> B.
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>
check sip.conf:

incominglimit=1
outgoinglimit=1