Asterisk Community

Hey Mikhail, saw your question in the group about VoIP platforms for ElevenLabs AI agents. ElevenLabs integrates perfectly with Twilio and Telnyx via SIP trunks for outbound routing. I actually provision premium Twilio and Telnyx infrastructure specifically configured for high-concurrency outbound AI pipelines (21-channel support) across the US, UK, and Australia. Because carrier blocking is a headache for AI agents, we focus heavily on clean routing and instant activation to keep connection rates high. If you are setting up your agent today and want to test the latency and audio quality, I can set up a quick 5-minute live test trunk for you right now so you can see how it performs with ElevenLabs. Let me know!

Can you install Asterisk on my Debian server and fully integrate it with MagnusBilling, including the necessary configuration and setup, so that the system is basically ready to use and I only need to add my SIP trunks/providers and configure the routing? Is that possible?

currently every single dude sells sip trunk .etc

tbh most of them don't even know what codecs are — unless they use ai to research on that.

does everyone have the experience of the outgoing call is normal but cannot receiving incoming calls in the PBX system?

I have the Internet IP SIP Trunk and the port

using the port UDP 5060

extension using the registration - username & password

So, where you can't receive you incoming call to? From trunk to PBX or from PBX to softphone/hardphone?

If the provider SIP Trunk pass the incoming calls with only UDP 5061, and my PBX 0.0.0.0 udp only listening the UDP 5060, it could be the reason that my phone didn't rings?

Hello, I need help with an outbound CLI / Caller ID issue. Could an experienced Asterisk, MagnusBilling or SIP-routing specialist please send me a private message? I am still quite new to this whole VoIP setup. I run MagnusBilling 8 with Asterisk 20 / PJSIP and an IP-to-IP upstream trunk. I only use German caller IDs. For the first two days, everything worked without any problems. I was able to change the Caller ID in MagnusBilling when needed, and the selected number was displayed correctly to the people I called. Since today, this suddenly stopped working. The calls still connect and ring normally, but the configured Caller ID is no longer displayed correctly. Instead, the called party sees random foreign fallback numbers, for example greece, Polish or Slovenian numbers. I checked the Asterisk SIP trace carefully. My server is sending the CLI correctly via P-Asserted-Identity, for example: P-Asserted-Identity: sip:497xxxxxxxx@my-server My provider also asked me to test adding 00 before the Caller ID, so I tested: P-Asserted-Identity: sip:00497xxxxxxxx@my-server Both formats are clearly being sent from my Asterisk server. The calls receive normal responses such as 100 Trying, 180 Ringing, 183 Session Progress and sometimes 200 OK. There are no relevant SIP rejection errors. I also tested multiple Germany routes and get the same result. My current PJSIP settings include: send_pai = yes from_user = trunk username send_rpid = no Could somebody please help me understand whether there is still a correct local Asterisk/PJSIP setting I should check, or whether this is clearly an upstream/carrier CLI mapping or whitelist issue? Please DM me if you can help. Thank you.

If anyone can help me with this issue, I would really appreciate a private message. I’m open to long term knowledge sharing, cooperation and helping each other both on a friendly and professional basis. Thank you guys 🙏🤝

Anyone need 3CX setup or SIP? Hit me up

the chan_sip removal thread got me paranoid, finally moved my last box to pjsip and ngl it was less painful than i expected