[asterisk-users] I loose incoming call after transfer

[jonas kellens]

Hello list.

An incoming call goes to the queue. Then is routed to a free
SIP-member1. When this SIP-member1 transfers the call to another
SIP-member2, and this SIPmember-2 rejects the call, then the
communication is lost.

How can I make the call go back to the SIP-member1 ? Or maybe back to
the queue ?

To transfer we use the 'transfer'-button on the Grandstream/YeaLink
IP-phone.

Greetingz.

Jonas.
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