* 3: Queue join. This will connect to the available agent. But to update the agent's status to 'available' it need manual API calling. So, this will not connect to the agent. Instead of connect to the agent, your call will stay at the queue's waiting actions which loop the given action. You can try how the waiting action is working.
* 4: Voicemail. You can leave some voice message. This is not special action but consists of some basic actions including the recording.
* 5: Conference. You will connect to the conference room. Currently, the conference room is open for anyone who pressed 4. So, you can talk to someone if someone in the conference.
* 6: Chatbot talk. You can talk with the chatbot(chatGPT). Btw, this will not work as your expectation. This is a sad story, to handle the RTP stream correctly, it costs a lot. So, I couldn't put many resources for this feature. So, this will working very slooooooooowly. And the RTP will have a loooooot of jitters. So, please don't expect the quality from here, and consider it is working if it saying something. Because that means it understood something.
* 0: Calling to project developer(me). - Not sure I wll answer. May will ignore the call.
Text to: +14703298699
"call me": Will make a call to you with simple talk action.
"text me": Will send a short SMS to you.
Note: ChatGPT 와 Transcribe 테스트는 생각만큼 잘 작동하지 않습니다. 영세하게 테스트환경을 꾸미다 보니, 충분한 리소스를 할당할 수 없었습니다. 게다가 서버가 유럽(gcp EU-zone)에 있다보니 거리에 따른 latency 지연도 상당합니다. 이점 양해해 주시고, 그저 뭔가 응답이 있으면 그냥 잘 작동하는구나 라고 봐주시면 감사하겠습니다. ^^;;
어쩌다보니 PSTN 이야기만 했네요. 하지만 당연하겠지만 WebRTC 역시 지원을 합니다. WebRTC(정확히는 JSSIP)을 이용하여 Registration 및 Trunking Call 이 가능하며, SIP/WebRTC/PSTN 과의 컨퍼런스 콜(Video/Voice mix) 역시 가능합니다.
다름이 아니고, 혼자서 진행하는 프로젝트이다 보니 피드백을 받기가 무척이나 어렵습니다. 한번 둘러보시고 프로젝트에 대해 작은 피드백 하나 부탁드립니다.
