Facing Issue With project Setup

[RashmiTechCraft]

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?

[danjenkins]

Are you also running the ARI project too? You need that too

[RashmiTechCraft]

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

[danjenkins]

Ah great ok.

See if you can send a netcat udp test from your asterisk box to your box handling the audio server?

[RashmiTechCraft]

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'
}```

[danjenkins]

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. (/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' }```

— Reply to this email directly, view it on GitHub https://github.com/nimbleape/asterisk-dialogflow-rtp-audioserver/issues/38#issuecomment-2136491708, or unsubscribe https://github.com/notifications/unsubscribe-auth/AAB3LLPK4G2UO5HASBHZP2LZEVKWXAVCNFSM6AAAAABIMV2CTOVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMZDCMZWGQ4TCNZQHA . You are receiving this because you commented.Message ID: @.*** .com>

[RashmiTechCraft]

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. (/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' }``` — Reply to this email directly, view it on GitHub <#38 (comment)>, or unsubscribe https://github.com/notifications/unsubscribe-auth/AAB3LLPK4G2UO5HASBHZP2LZEVKWXAVCNFSM6AAAAABIMV2CTOVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMZDCMZWGQ4TCNZQHA . You are receiving this because you commented.Message ID: @. .com>

I am using FreePBX Setup

Here is the Bridge which is inside same machie as freepbx machine

yarn run v1.22.22
$ node index.js
{"level":30,"time":1716979190779,"pid":2519971,"hostname":"ip-172-31-54-247","name":"Asterisk-Dialogflow-ARI-Bridge","msg":"Starting","v":1}
{"level":30,"time":1716979190781,"pid":2519971,"hostname":"ip-172-31-54-247","name":"Asterisk-Dialogflow-ARI-Bridge","msg":"trying to connect to mqtt","v":1}
{"level":30,"time":1716979191064,"pid":2519971,"hostname":"ip-172-31-54-247","name":"Asterisk-Dialogflow-ARI-Bridge","msg":"connected to mqtt","v":1}
{"level":30,"time":1716979191065,"pid":2519971,"hostname":"ip-172-31-54-247","name":"Asterisk-Dialogflow-ARI-Bridge","ariConfig":{"url":"localhost:5038","username":"admin","password":"ASTERISK_PASS","appName":"dialogflow"},"msg":"ari config","v":1}

And Here is the Log Of Audio Server That is running inside a separate VM

root@ip-172-31-23-32:/home/asterisk-dialogflow-rtp-audioserver# yarn start
yarn run v1.22.22
$ node index.js
{"level":30,"time":1716979571344,"pid":1730,"hostname":"ip-172-31-23-32","name":"Dialogflow-AudioServer","msg":"Connected to MQTT","v":1}
{"level":30,"time":1716979571513,"pid":1730,"hostname":"ip-172-31-23-32","name":"Dialogflow-AudioServer","msg":"Subscribed to both newStream & streamEnded topic","v":1}
{"level":30,"time":1716979571515,"pid":1730,"hostname":"ip-172-31-23-32","name":"Dialogflow-AudioServer","msg":"AudioServer listening on UDP port","v":1}

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?

[RashmiTechCraft]

@danjenkins Please let me know the steps to make sure the calls passes on to the ARI bridge rather than extension. Thanks

[danjenkins]

You need to add the Stasis application into the call.

https://community.freepbx.org/t/add-stasis-application-in-dialplan-to-use-asterisk-ari-with-freepbx/49792

Thats how ARI works - replace Stasis(hello-world) with the name of the ari app youve put into the config