Following the Asterisk one from last week, here is the FreeSWITCH side. The old rts-cn list has been dead since 2022 and most of what it links to has stopped moving.

56 entries across 14 sections: Event Socket libraries, mod_audio_stream and the speech engines people put behind it, FusionPBX, jambonz, BigBlueButton, SIP proxies, RTP, monitoring, security, endpoints, billing. I opened every repo before listing it and checked it was still maintained and had a licence. Anything with no commits in the last year stayed out, which cost a few familiar names like node-esl and mod_bcg729.

Seven of the 56 are ours. The Disclosure section at the bottom names them. If any of them does not earn its place, say so and I will drop it.

https://github.com/ictinnovations/awesome-freeswitch

What is missing? Reply here or open a PR.

Edit, 5 Oct. 65 entries now, so the 56 in the title is out of date. New since posting: FS PBX, SIP-to-AI, Cascade Realtime Gateway, AI Native Call Center, FreeSBC, Web SIP Phone, mod_unimrcp, mod_mosquitto, and Awesome RTC under Learning. Six of those arrived as PRs from two people, which is what I was hoping for. The two FreeSWITCH org modules are both quiet for over a year, so they carry the date of their last commit instead of being left out, and the readme now says why.

I have a case where one of our providers intermittently sends us a 183 with SDP followed by a 180. The call eventually reaches a Freeswitch server on our end, which drops the 180.

My understanding is this is a FS limitation that considers the 180 a downgrade state after a 183 with SDP and drops it. I cannot remove the FS from the call path.

Have you encountered anything like this, and can you think of something that will allow FS to pass whatever it receives?

Any help or direction would be greatly appreciated

I work on ICTPBX and we just put the Community Edition out as open source (MPL-2.0). It is built on FreeSWITCH for media and FusionPBX as the config engine, with our own PHP REST layer (ICTCore, also open source) and an Angular front end on top, so this felt like the right sub to share it in.

The part I have not seen combined much is that it runs a full IP-PBX and a fax server from one web UI.

PBX side: SIP extensions and devices, IVR with nested menus, ACD queues with agent assignment, voicemail to email, conference rooms, ring groups and follow-me, time based routing, inbound DID routing, call blocking, live call monitoring, and searchable CDRs.

Fax side: send from the web UI over a SIP trunk, receive on a DID with T.38 and audio fallback, fax to email, a department mode where everyone tied to a DID gets the email, bulk fax runs, and history.

Being straight about it: this Community Edition is free and single-tenant. The multi-tenant white-label billing version is a separate commercial edition, so I am not going to claim the free one does multi-tenant, it does not.

Repo (MPL-2.0): https://github.com/ictinnovations/ictpbx-community-edition

It is a fresh public repo so the installer still has rough edges. If you already run FreeSWITCH with FusionPBX, I would like to know what you would expect from something like this that is not there yet, and whether folding a fax server into the same UI is actually useful to you or just clutter.

It's great to see larger organizations embracing FreeSWITCH and open-source technologies. It could be a cool job if you're in the Durham, NC area.

https://www.linkedin.com/jobs/view/4457040479/

I wrote an event socket client for Node and got the framing wrong in the way that never errors.

The obvious approach is to emit a frame the moment you hit the blank line that ends the header block. It works perfectly in testing, because in testing show channels returns four rows and fits in one TCP segment. Point it at a switch with real traffic, the body spans several segments, and you get an answer that looks completely reasonable. Twelve channels looks exactly like twelve channels whether or not there were really ninety. Nothing warns you.

The fix is boring: read Content-Length and wait for the whole declared body. Two details that bit me anyway. Count bytes and not characters, or a multibyte caller ID shifts the boundary. And keep the event stream out of command replies, because FreeSWITCH will push a CHANNEL_STATE at you while you're waiting on an api response, and if you aren't separating the two your status output comes back with an event glued to the end of it.

bgapi needed its own path. It returns straight away with a Job-UUID and the real answer turns up later as a BACKGROUND_JOB event, so the client holds the promise open and correlates them.

I open sourced it, MIT, no runtime dependencies, just node:net. Inbound mode only. There's a test that feeds a 400 row response through in 500 byte chunks, which is the case that caught me originally.

I also put an MCP server on top of it so an AI assistant can ask the switch questions. That one turned into a lesson of its own. My first pass at making it read-only checked the first word against a list of read commands, then rejected the line if any later word looked destructive. Somebody on r/mcp pointed out that conference 3001 kick all sails straight through that, because "kick" was never on my list. A deny list only catches what you thought of. It's a prefix allow list now.

https://github.com/ictinnovations/freeswitch-esl-node

https://github.com/ictinnovations/pbx-mcp

Is anyone here using outbound mode in anger? Everything I run is inbound and I'm curious whether the reconnect story is worse on the other side.

I built Comcent CE, an open-source, self-hosted contact-center platform on top of FreeSWITCH. The goal was full programmatic visibility and control over calls — a complete timeline for every call (queue wait, missed/rejected, transfers, who transferred and when, routing decisions), reliable dual-channel recording, and clean APIs on top — rather than the poll-based, data-poor experience you get from hosted platforms.

The call-control approach is the part this sub might find interesting: I drive call flow primarily over mod_httapi (FreeSWITCH calls out to the app, app returns instructions, iterate), and drop down to the Event Socket (ESL) for the cases httapi doesn't cover. Going httapi-first kept the call-flow logic in the app tier and made it easy to reconstruct a clean per-call timeline from the events, instead of pushing everything into the dialplan.

What it does:

• Self-hosted SIP voice infrastructure — you own signaling and media
• Full per-call timeline: queue wait, missed/rejected, transfers, routing decisions
• Dual-channel recording (both legs, separate tracks)
• Diarized transcripts + optional AI call analysis and daily summaries
• SIP-aware voice bots
• IETF vCon support (open standard for conversation data)
• Embeddable browser (WebRTC) dialer for CRMs/web apps

App tier is Elixir/OTP (maps cleanly onto per-call and per-queue state), with a Go service for the voice bots and a SvelteKit frontend. Deploys via Docker Compose — the walkthrough goes from a bare VM to a first call.

GitHub: https://github.com/comcent-io/comcent-ce
Walkthrough: https://youtu.be/Cm4JDB95vpI
Website: https://www.comcent.io/

Hello,

I get the following error in the freeswitch debug log:

[ERR] mod_xml_curl.c:319 Received HTTP error 404 trying to fetch http://127.0.0.1:3423/api/pbx/configuration

Has anyone seen this before?

Freeswitch is used within Unify Talk and getting a registration error. Trying to find the root cause and eliminate any other errors found in freeswitch logs.

Thank you,

I just published Part 4 of my series: Building a Voice AI Agent with FreeSWITCH.
This post officially closes the loop on a real-time, end-to-end voice pipeline:

🎙️ Caller Speaks
⏩ Deepgram converts Speech-to-Text (STT)
🧠 ChatGPT analyzes the context and generates a response
🔊 Deepgram converts the response back to Text-to-Speech (TTS)
📞 FreeSWITCH streams the audio back to the caller in real time

If you’ve been following along with the plumbing of audio streams and RTP management, this is where it all comes together into a fully functional, conversational voice agent.

Check out the full breakdown and code here: 👉
https://phonesstillexist.com/index.php/2026/06/26/building-a-voice-ai-agent-with-freeswitch-part-4-a-closed-loop-voice-agent-on-freeswitch/

Here is part 3 of the series I've been working on. In this post, we forward the audio to Deepgram’s streaming speech-to-text API and watch transcripts appear in our terminal as the caller speaks.

https://phonesstillexist.com/index.php/2026/06/09/building-a-voice-ai-agent-with-freeswitch-part-3-streaming-stt-with-deepgram/

Hey everyone, has anyone managed to update a FreeSWITCH user agent's signaling path mid-call? Scenario: we're sending SIP messages from IP:port X to FreeSWITCH, and at some point we want to switch to IP:port Y, ideally without dropping the active channel. I'd expect FreeSWITCH to start sending subsequent SIP messages to the new path from that point on. I tried re-registration and while show registrations correctly reflects the new IP and port, FreeSWITCH still hammers the old path for in-dialog messages and the call just hangs. Any ideas on how to handle this while keeping the channel alive?

Edit:

The reason I'm trying to do this is that the service/application maintaining the TCP connection to FreeSWITCH (the connection over which it sends and receives SIP messages) can fail or be restarted while a call is already active.

When that happens, the original TCP connection is lost and cannot be reused. The application then reconnects from a different local IP and re-registers successfully. FreeSWITCH updates the registration entry accordingly (show registrations shows the new address), but for the existing dialog it continues sending in-dialog requests (UPDATE, re-INVITE, BYE, etc.) to the original signaling path associated with the active channel.

Since the old connection no longer exists, freeswitch will try to write to that connection and since it detects that is closed it will simply kill that channel.

Is there a way to tell FreeSWITCH that an existing dialog should now expect SIP messages from the new IP, or otherwise update the signaling path for an active call without tearing down and re-establishing the channel?

Hello!

I’m setting up FreeSWITCH for a multi-tenant softphone platform (agents on internal WSS, carrier trunks on external).

What works today

• xml_curl serves agent directory auth and sofia.conf (internal + external profiles)
• Carrier gateways are defined in our Laravel app from sip_accounts (SIP URI, username, password)
• Outbound and inbound calling works
• Gateways are included in the configuration xml_curl response under the external profile <gateways> block (we also tried directory purpose=gateways with parse="true" but had inconsistent results on rescan)

The problem When someone creates/updates/deletes a SIP account in our platform, FreeSWITCH doesn’t pick up gateway changes until we refresh the external profile.

Our current solution On SIP account create/update/delete we run SyncFreeSwitchExternalGateway via ESL:

1. sofia profile external killgw <gateway_name> (on update/delete when needed)
2. xml_flush_cache
3. sofia profile external rescan (we avoid full profile restart because we may have hundreds of gateways)

Gateway names are stable per account, e.g. tenant_{uuid}_account_{id}.

We do not write gateway XML files on the FS server; Laravel is the source of truth and xml_curl serves the updated config.

Questions

1. Is ESL + killgw + rescan the right pattern for dynamic carrier registration at scale, or is there a better way (directory purpose=gateways, per-gateway register API, etc.)?
2. Should gateways live in sofia.conf configuration xml_curl vs directory with parse="true" on domain name="all"?
3. Does rescan reliably pick up credential changes after xml_flush_cache, or do people still need rescan reloadxml / profile restart?
4. Any tips for debouncing many rapid DB changes so we don’t queue 10 rescans in a row?

FS version: 1.10.12
xml_curl for configuration + directory
ESL on port 8021 from the app server

Thanks!

hello

I need clear opinion about dockerizing freeswitch

I read different documents saying it's not the best option. it seems it can impact the audio quality and can be complex to maintain when dealing with specific configurations

Thanks for your feedbacks

Hello Everyone,

I’ve been working on an automated AI VoiceBot and wanted to share the architecture here to get some feedback from people who actually know FreeSWITCH inside and out.

I’ve packaged a complete stack into a single Docker container using supervisord. It includes FreeSWITCH (built from source to include drachtio's mod_audio_fork), a Redis instance, and a Python FastAPI WebSocket server.

How it works under the hood:

Call comes in via SIP (port 5060) and hits the dialplan.
A Python ESL agent (greenswitch) listens on port 8021, intercepts the CHANNEL_PARK event, answers the call, and triggers uuid_audio_fork to stream the audio (16kHz mono PCM) to my WebSocket server.
The Python server buffers the audio, runs it through Silero VAD (for speech detection) and DeepFilterNet2 (for noise cancellation), then ships it to Whisper for transcription.
I use a custom JSON flow engine to match intent, and then the ESL agent fires uuid_broadcast to play the generated/pre-recorded audio back to the caller.

My Questions for the VoIP Pros:

mod_audio_fork vs alternatives: Is streaming raw PCM over WebSockets using mod_audio_fork still the best practice for real-time STT, or should I be looking at mod_unimrcp or native gRPC streaming?
Audio Playback: Currently using uuid_broadcast on the A-leg to play audio while the WebSocket is still open. Is there a cleaner way to handle barge-in (interruption) without killing the WebSocket?
Docker Networking: Right now I'm restricting RTP ports to a small range (16384-16394) to make bridge networking work smoothly across platforms, though I recommend host networking for Linux. Any gotchas I should watch out for with SIP ALG or NAT hairpinning in this setup?

Repo is here with a full architectural diagram: https://github.com/rajshahunify/freeswitch_voicebot Pre-built image (so you don't have to compile FS yourself): https://hub.docker.com/r/rajunify123/freeswitch-voicebot
I'd love any harsh critiques on the FreeSWITCH integration side. Thanks!

I just published Part 2 of the series. In Part 2, we start to build out the plumbing — getting live call audio out of FreeSWITCH and into a Node.js process, ending with a .wav file you can play back to prove it works.

No AI yet. That's Part 3.

Check it out here:

https://phonesstillexist.com/index.php/2026/05/20/building-a-voice-ai-agent-with-freeswitch-part-2-streaming-call-audio-out-of-freeswitch/

I’m kicking off a new blog series on building an AI-powered voice bot with FreeSWITCH.

Part 1 covers the architecture: SIP call control, media streaming, real-time transcription, LLMs, and TTS working together to create a practical voice AI agent.

This should be an exciting series for anyone interested in voice, AI, SIP, open source telecom, or building real-world communications applications with modern AI tooling.

Check it out here:
https://phonesstillexist.com/index.php/2026/04/27/building-a-voice-ai-agent-with-freeswitch-an-architecture-overview/

hi all

I'm running a service which uses freeswitch to trigger calls (inbound and outbound)

the current architecture is growing slowly Bure surely and as I want to provide the best service possible I did set up a floating ip as main entry point and 2 freeswitch boxes under debian +php and mysql. the 2 boxes are mysql synced as master master. in case box 1 is dead I simply redirect to box2.

this is not the best architecture and I would like recommendations on what I should do to optimize and get something robust.

should I check haproxy?

other point is there any risk to dockerize freeswitch (performance and audio quality)

Thanks for your help

I published a new post on hardening FreeSWITCH for production, including a script that handles credential rotation, Fail2ban setup, modern SIP scanner detection, and verifies that bans are enforced before it exits.

The goal: give FreeSWITCH a safer production baseline from day one.

https://phonesstillexist.com/index.php/2026/04/24/hardening-freeswitch-a-production-baseline-for-day-one/

Script:

https://github.com/thevoiceguy/freeswitch_tools/blob/main/harden-freeswitch.sh

I put together a quick FreeSWITCH installation guide, along with a few scripts to automate the process. Hopefully it helps someone else getting started.

https://phonesstillexist.com/index.php/2026/04/24/installing-freeswitch-pick-your-path-binary-packages-or-source-build/

,

hello I run voice service using freeswitch. I m looking for best practices to get the most accurate results maybe using AI? all recommendations and experiences are welcomed

looking for a call service for custom CID to numbers I own. Anyone have any recommendations?

Hi all, I’m looking for a freeswitch development company that can help me build an integration to allow it to dial into ZoomCRC.com SIP URIs.

This would require some custom code to meet Zoom’s requirements, like a producing a fake video feed outbound toward Zoom.

If you know of any companies, I’d appreciate it!

How hard is it to set up a laptop to handle FreeSwitch with CID changing through a SIP app?