[stream-audio]
; Handle calls between any two extensions in the 7000 range using external media
exten => _7XXX,1,NoOp(Starting call to ${EXTEN} from ${CALLERID(num)})
same => n,ExternalMedia(rtp,3.252.149.9:3000/3.252.149.9:3001,slin16) ; Send RTP to your Node.js server ports
same => n,Hangup()
; Voicemail handling for any extensions in the 8000 range
exten => _8XXX,1,VoicemailMain(${EXTEN}@main)
same => n,Hangup()
Well, ExternalMedia is not a dialplan application. You use it in ARI. There’s been quite a few threads on it, which you can find by searching or someone else may chime in. How did you come up with the given dialplan?
I have configured the call from the extensions.conf, I succeeded in making the record on the asterisk side with the mixmonitor but I want to know how can I listen in the real time to the call using external media from the channels Rest API?
What configurations should I have to make in the extensions.conf?
[stream-audio]
; Handle calls between any two extensions in the 7000 range using external media
exten => _7XXX,1,NoOp(Starting call to ${EXTEN} from ${CALLERID(num)})
same => n,ExternalMedia(rtp,3.252.149.9:3000/3.252.149.9:3001,slin16) ; Send RTP to your Node.js server ports
same => n,Hangup()
; Voicemail handling for any extensions in the 8000 range
exten => _8XXX,1,VoicemailMain(${EXTEN}@main)
same => n,Hangup()
You would call the ARI application using the Stasis dialplan application, and specifying the name of the ARI application. It is up to you to write the ARI application though.
it should be in the extensions.conf:
[stream-audio]
; Handle calls between any two extensions in the 7000 range using external media
exten => _7XXX,1,NoOp(Starting call to ${EXTEN} from ${CALLERID(num)})
same => n,ExternalMedia(rtp,3.252.149.9:3000/3.252.149.9:3001,slin16) ; Send RTP to your Node.js server ports
same => n,Hangup()
; Voicemail handling for any extensions in the 8000 range
exten => _8XXX,1,VoicemailMain(${EXTEN}@main)
same => n,Hangup()
the answer that I got it, it is not what I need, and I don’t understand why are you talking in an impolite way, my question was clear, I want to make real-time connection from my node js code to listen in the real-time to the call, and make the transcription of the call.
all that I need are the configurations from the asterisk side??? @david551
"Yes, sure! I successfully managed to record the call using MixMonitor. I stored the recording on the server and then retrieved it using Asterisk’s API via /recordings/stored.
Now, I’m trying to enable real-time listening of the call to perform transcription. What I need to know is what specific configurations I should add to the extensions.conf file to achieve this."
Feel free to come back with any questions once you knwo what ARI is and what
it’s for.
Antony.
–
The words “e pluribus unum” on the Great Seal of the United States are from a
poem by Virgil entitled “Moretum”, which is about cheese and garlic salad
dressing.
–
Never automate fully anything that does not have a manual override capability.
Never design anything that cannot work under degraded conditions in emergency.