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Facing Issue With project Setup #38
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Are you also running the ARI project too? You need that too |
Yes I am Running the ARI Project On The Same Machine As Asterisk |
Ah great ok. See if you can send a netcat udp test from your asterisk box to your box handling the audio server? |
Yes While Doing The Below Command I Am getting ping inside audio server
|
Great. You’ll need to enable more logging in asterisk to see what’s
happening with the rtp then.
In the asterisk cli go and enable rtp logging and see what’s happening to
the rtp
*Dan Jenkins*
Founder @ Nimble Ape / Everycast Labs / CommCon
Nimble Ape: nimblea.pe
Everycast Labs: everycastlabs.uk // broadcastbridge.app
CommCon: commcon.xyz
…On Wed, 29 May 2024 at 05:30, RashmiTechCraft ***@***.***> wrote:
Ah great ok.
See if you can send a netcat udp test from your asterisk box to your box
handling the audio server?
Yes While Doing The Below Command I Am getting ping inside audio server
echo "Hello from WSL" | nc -u MY_SERVER_IP 7777
{"level":30,"time":1716956739128,"pid":1660,"hostname":"ip-172-31-23-32","name":"Dialogflow-AudioServer","msg":"Subscribed to both newStream & streamEnded topic","v":1}
{"level":30,"time":1716956739129,"pid":1660,"hostname":"ip-172-31-23-32","name":"Dialogflow-AudioServer","msg":"AudioServer listening on UDP port","v":1}
node:buffer:1168
throw new ERR_INVALID_BUFFER_SIZE('16-bits');
^
RangeError [ERR_INVALID_BUFFER_SIZE]: Buffer size must be a multiple of 16-bits
at Buffer.swap16 (node:buffer:1168:11)
at Socket.<anonymous> (/home/asterisk-dialogflow-rtp-audioserver/lib/RTPServer.js:47:21)
at Socket.emit (node:events:519:28)
at UDP.onMessage [as onmessage] (node:dgram:943:8) {
code: 'ERR_INVALID_BUFFER_SIZE'
}```
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I am using FreePBX Setup Here is the Bridge which is inside same machie as freepbx machine
And Here is the Log Of Audio Server That is running inside a separate VM
Now Whenever i am placing a call to my Twilio Phone, I am Getting the same call Inside My VOIP Softphone and the conversation continues but i do not see extra logs in both The ARI Bridge Or Audio Server Is there any Settings that i am missing here? |
@danjenkins Please let me know the steps to make sure the calls passes on to the ARI bridge rather than extension. Thanks |
You need to add the Stasis application into the call. Thats how ARI works - replace Stasis(hello-world) with the name of the ari app youve put into the config |
I am Trying to setup the project and seems like i am not getting any logs
{"level":30,"time":1716892244292,"pid":1,"hostname":"bbe63e60c3ec","name":"Dialogflow-AudioServer","msg":"Connected to MQTT","v":1} 2024-05-28T10:30:45.223190860Z {"level":30,"time":1716892245222,"pid":1,"hostname":"bbe63e60c3ec","name":"Dialogflow-AudioServer","msg":"Subscribed to both newStream & streamEnded topic","v":1} 2024-05-28T10:30:45.223906037Z {"level":30,"time":1716892245223,"pid":1,"hostname":"bbe63e60c3ec","name":"Dialogflow-AudioServer","msg":"AudioServer listening on UDP port","v":1}
Above is the only 3 line log i am getting nothing more.
is this a port issue? Should i keep it in the same subnet or same machine as asterisk or different server?
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