Victor Villarreal

Dear @airween , thanks for your feedback and sorry for the delay. I'll check with the last version of ModSecurity if the risk indicated on the website is still valid...

Hi @UmairShah7677, I don't see that Asterisk is sending the extension number when Dial. You can try this: `Dial(SIP/[email protected])` where `EXT_NUMBER` is the number that you configured in Livekit Cloud...

Hi @ccameroVzy, I've read that Livekit Cloud, only works with TCP packets. Not UDP. Try with TCP communication.

@UmairShah7677 , I think that your Asterisk needs to communicate with Livekit Cloud via TCP. To do so, follow this article: https://support.digium.com/s/article/How-to-use-SIP-over-TCP Note that Asterisk 1.8 and above support SIP...

Hi. I think a VoIP traffic trace would be helpful. Maybe a sngrep trace? I understand that the call will keep ringing until the first DSP (handshake) packet arrives 🤷🏻‍♂️

Hi @stiven77nj 👋🏻 The config looks ok for me. Please, take into account that SDP media packets may came from a host with an IP distinct that the SIP signalling...

Hola @stiven77nj 👋🏻 Por el mensaje del servidor/contenedor livekit, pareciera que no existe un inbound trunk que coincida con la llamada entrante que le estás enviando. También debes crear una...

Great! Thank you for allowing me to collaborate on this wonderful project ❤️.