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Hi, I have missed something so I wonder if someone could explain for me? 0424449647 desktop phone 0106024647 DECT phone 0424449630 Helsingborg queue extensions.conf --------------- [support] exten => 0424449647,hint,SIP/0424449647&SIP/0106024647 exten => 0424449647,1,Dial(SIP/0424449647&SIP/0106024647,15,rtT) [inputinterior.se] exten => 0/0424449647,1,Answer() exten =>

I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their

...dded>Raised when a member is added to the queue). My idea is to send the event with the help of a script via AMI. First i tried to do this manually connected to the AMI via telnet and pasted it in: Event: QueueMemberAdded Privilege: agent,all Queue: 905 MemberName: SIP/599 Interface: SIP/599 StateInterface: SIP/599 Membership: realtime Penalty: 0 CallsTaken: 0 LastCall: 0 LastPause: 0 InCall: 0 Status: 1 Paused: 0 PausedReason: 0 Ringinuse: 0 This generates an error: Response: Error Message: Missing action in request I wasn´t able to find the correct action for sending events. Should this action l...

...the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet: Action: QueueAdd Queue: queuetest MemberName: 1234 Interface: PJSIP/6001 StateInterface: PJSIP/6001 Ringinuse: yes Paused: false If I change to SIP, the extension will call normally. My configuration pjsip.conf [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0:5160 local_net=192.0.0.0/24 external_media_address=192.168.0.196 external_signaling_address=192.168.0.196 [6001...