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

[Brian Capouch]

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.