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I will fix voip sip trunk and connect ai voice agent on asterisk freepbx 3cx
Software Developer
When a client tells me their team wastes hours on repetitive tasks, their
documents are impossible to search, or their SaaS idea has been stuck in
their head for months I know exactly how to fix that....
Informazioni su questo servizio
Your AI voice agent works in the demo. Then it hits a real phone line and breaks.
Silent calls. One-way audio. Calls that die at 32 seconds because the ACK never arrived. Transfers that go nowhere. That is not the AI failing. It is the SIP layer under it.
I fix that layer on any VoIP PBX, and build it from scratch when nothing is there.
WORKS WITH
Vapi, Retell, Bland, ElevenLabs, LiveKit, Pipecat, OpenAI Realtime
Asterisk, FreePBX, 3CX, FusionPBX, FreeSWITCH
Twilio, Telnyx, SignalWire, Plivo, or your own carrier
YOU GET
- Your AI voice agent live on your own phone number or SIP trunk
- Inbound and outbound routing that holds under load
- Warm transfer to a real human that actually completes
- Recording, keypad presses and a menu fallback if the agent fails
- A recorded test call as proof, with jitter and packet loss measured
TO START I NEED
Your agent platform, your trunk or PBX access, and what a good call looks like. Credentials stay in this order thread, are used only for your build, and you rotate them the day I deliver.
IF IT CANNOT BE DONE
I tell you in the first hour with the SIP trace that proves it.
Message me before ordering. I reply in under an hour.
Provider:
Cisco
•
VICIdial
•
3CX
•
RingCentral
•
Asterisk
•
FreePBX
•
twilio
Il mio portfolio
FAQ
Do you include Vapi, Retell, Twilio or SIP trunk subscription fees?
No. You pay Vapi, Retell, Twilio, Telnyx or your carrier directly, so the accounts and numbers stay in your name. My fee covers diagnosis, configuration and testing only. I never put your telephony inside an account you cannot move or cancel yourself.
My agent answers but transfers to a human fail. Can you fix that?
Yes, and it is the most common job I get. Failed warm transfers usually trace to REFER handling, a re-INVITE your carrier rejects, or a missing Diversion header. I reproduce it on a live call, read the SIP trace, then fix the leg that is actually breaking.
One way audio after the agent picks up. Is that in scope?
Yes. One way audio, choppy or robotic speech, the agent talking over the caller, and DTMF keypresses that never register are all in scope. These usually trace to RTP or NAT paths, codec mismatch, or the wrong DTMF mode (RFC2833, inband or SIP INFO).
Do you do FreePBX and Asterisk PBX setup from scratch?
Yes. If you have no PBX yet I install and configure FreePBX 17 or Asterisk on your server or VPS, then connect the SIP trunk and your AI voice agent. Choose Standard or Premium for a from scratch build. Basic assumes you already have a working PBX.
What if the problem turns out to be my carrier and not my setup?
You still get the answer in writing. I send the SIP trace, the failing call leg, and a message you can forward straight to your provider. If nothing on your side can fix it, I say so and we cancel rather than bill you for a problem that is not yours.
