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.
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
check sip.conf:
incominglimit=1
outgoinglimit=1
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
