Hi ,
The SIP agents get logged out after transfer the call through transfer button not through dtmf ( # ) . Is it possible to avoid in asterisk1.4 .
Thanks
Hi ,
The SIP agents get logged out after transfer the call through transfer button not through dtmf ( # ) . Is it possible to avoid in asterisk1.4 .
Thanks
The agent isn’t doing the transfer, it is the agent’s phone, outside the queuing and agent system, so this is not possible in any version of Asterisk.
It is of the nature of SIP transfers that the original channel to the transferring phone gets hung up.