This application allows users to dial any sip extension and ask any question from Gemini

git clone https://github.com/aqeelabpro/gemini-asterisk

cd gemini-asterisk

open in your favourite editor or IDE

• FreePBX installed
• Zoiper Installed

• Clone this repository https://github.com/aqeelabpro/freepbx-16-ubuntu-20.04-installation
• cd freepbx-16-ubuntu-20.04-installation
• chmod +x freepbx_16_asterisk_18_install_ubuntu_20.04.sh
• ./freepbx_16_asterisk_18_install_ubuntu_20.04.sh

• Goto Config Edit in the Admin tab in FreePBX(you will have to download config edit module)
• Paste the code provided at the end to the extensions_custom.conf file, and click Save button and the Apply Config button at the top
• create Jar file from this Application and upload to server
• Create a main folder like gemini-asterisk
• Inside gemini-asterisk folder, create 2 folders, config and bin
• Inside the config folder, create 2 files application,yml and paste content from this Java Application
• Inside the Same config folder, create another file log4j2.xml and paste log4j2.xml from this Java Application
• Create a log folder, let's say the main folder gemini-asterisk is in /home/ubuntu, create log folder in same /home/ubuntu

• Goto Admin Tab as shown in Figure.1
• Select the Updates from dropdown as shown in Figure 1.1
• Click The Modules Updates as shown in Figure 1.2
• Click Standard and Extended in there and hit check online button as shown in Figure 1.2
• You will get Download All button, Press that button and it will download all modules including Config Edit module, we require as shown in Figure 1.2

Figure 1.1

Figure 1.2

[google-speech]
exten => 111,1,NoOp(============ ${CONTEXT} =============)
exten => 111,n,Set(__AGI_SERVER_IP=127.0.0.1)
exten => 111,n,Set(__AGI_SERVER_PORT=9000)
exten => 111,n,Set(__AGI_SERVER=agi://${AGI_SERVER_IP}:${AGI_SERVER_PORT})
exten => 111,n,Set(__SOUND_FOLDER=/var/lib/asterisk/sounds/en)
exten => 111,n,Set(__LANG_CODE=en-GB)
exten => 111,n,Set(__LANG_NAME=en-GB-Wavenet-F)
exten => 111,n,Goto(record-answer,111,1)

exten => h,1,Goto(grace_fully_hangup,s,1)

[record-answer]
exten => 111,1,NoOp(=========== ${CONTEXT} ===========)
exten => 111,n,Set(__RECORDING_FILE_NAME=${EPOCH})
; ADDED THIS LINE
exten => 111,n,Playback(ask_question&interrupt-ai)
exten => 111,n(record),Record(${SOUND_FOLDER}/${RECORDING_FILE_NAME}:wav,8,300)
exten => 111,n,AGI(${AGI_SERVER}/speechToText.agi)
exten => 111,n,Goto(play-prompt,111,1)

exten => h,1,Goto(grace_fully_hangup,s,1)

[play-prompt]
exten => 111,1,NoOp(=========== ${CONTEXT} ===========)
exten => 111,n,AGI(${AGI_SERVER}/textToSpeech.agi)
exten => 111,n,Wait(1)
exten => 111,n,Background(${SOUND_FOLDER}/${ANSWER_FILE})
exten => 111,n,Goto(record-answer,${EXTEN},record)

exten => _X,1,Goto(record-answer,111,record)

exten => h,1,Goto(grace_fully_hangup,s,1)

[grace_fully_hangup]
exten => s,1,NoOp(=========== ${CONTEXT} ===========)
exten => s,n,StopMonitor()
exten => s,n,Hangup

Just install Zoiper and Register a SIP user

• SIP URI: 2000@34.125.134.112
• Password: 94d6a19791656fc64f4e55f7867fc407

After Registering The Account, dial extension 111, and follow the instructions in the recording played during the call to ask any questions
it has some latency due to the conversion of speech-to-text and text-to-speech using Google