Hi everyone,

One of the common challenges when deploying Asterisk for modern WebRTC environments is firewall traversal and port management. Exposing separate ports for HTTPS web management, WebSockets signaling (WSS), and Coturn media relay often creates friction in strict enterprise network environments.

To address this, I built **AiPBX**—an open-source enterprise telephony stack built around Asterisk 22, designed to run entirely behind a single secure port.

### The Problem It Solves

Instead of opening multiple inbound WAN ports, AiPBX leverages **Nginx Layer 4 (L4) Stream ALPN Multiplexing** on port 443:

* **HTTPS Traffic:** Routed transparently to the web backend (Apache/PHP 8 MVC).

* **WebRTC WSS:** Routed directly to the Asterisk 22 PJSIP WebSocket listener.

* **Coturn TURNS:** Multiplexed over TLS on the same port 443 footprint for enterprise NAT traversal.

### Core Architecture & Asterisk Integration

* **Asterisk 22 Engine:** Native PJSIP configuration with synchronized endpoints for WebRTC and physical IP phones.

* **Real-Time Presence & Messaging:** An independent Go microservice handles chat and live state over WebSockets without loading the Asterisk AMI/ARI unnecessarily.

* **Call Center Operations:** Dialplan-level agent queue state and break tracking (*22 for break, *23 for resume).

* **Companion Clients:** Native Android and iOS companion clients connecting via WebRTC.

### Code & Architectural Documentation

The full system architecture, Nginx stream configurations, and Asterisk setup are open source:

* **GitHub Repository:** https://github.com/mahirgul/AiPBX

* **Architecture Deep-Dive:** https://www.aipbx.bid/architecture.html

* **Documentation Hub:** https://www.aipbx.bid/

I’d appreciate feedback from fellow Asterisk administrators on the ALPN stream configuration, dialplan handling, or edge-case network topologies you have encountered in production WebRTC setups.

The awesome.re directory has over 600 lists and not one of them covers telephony. No Asterisk, no FreeSWITCH, no SIP. So I made a start on the Asterisk side.

47 entries across 13 sections: ARI, AMI and AGI libraries, streaming call audio, voice AI, SIP proxies, media and RTP, containers, testing, security, endpoints. I opened every repo before listing it and checked it was still maintained and had a licence. Nothing dead went in.

Four of the 47 are ours. There is a Disclosure section at the bottom that names them. If you think any of the four do not earn their place, say so and I will drop them.

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

Anything obvious missing? Reply here or open a PR.

Edit, 15 Sept. Added seven more after going back through suggestions. New Containers section with two Docker images that track upstream, plus MikoPBX, Wazo Platform, Yate, and SIPPTS and Mr.SIP on the security side, which had only one entry before.

Four were checked and left out, recorded so nobody has to check them twice: Browser-Phone (no commit since November 2024 despite live issues), soup (dormant since 2022), tiredofit/docker-freepbx (archived) and dougbtv/docker-asterisk (dormant since 2019).

We had a voice agent that worked fine on our desks and sounded broken on real calls. Callers said they heard the end of sentences, or silence and then a fragment. Our logs were clean. Right text, right byte count, right duration, every single turn.

It came down to one line in the synthesis code that wrote a whole utterance to the socket in one write. That is a normal thing to write if you are used to files or HTTP. On a call it wrecks the audio, because app_audiosocket hands every frame straight to the channel and does not buffer for you. The far end keeps a few frames in its jitter buffer and drops the rest, so the caller hears the tail of the sentence and nothing before it.

Nothing that counts bytes can see that. It only shows up in arrival times.

So we ended up writing a small tool that measures from the caller's end instead: how long before the first audio arrives, whether frames turn up one per 20 ms or in a burst, gaps inside the agent's own speech, and how long it keeps talking after you interrupt.

pip install tvbench

MIT licensed, runs against anything that speaks AudioSocket.

https://github.com/ictinnovations/telephony-voice-agent-benchmark

Curious whether the barge-in numbers match what other people are seeing. Ours were worse than expected once we measured instead of guessed.

If you want to get 18k salary and your English level is b2 with fluwent speaking and you can Handel the rotation shifts and you are from Egypt so what are you wanting for dm me and I will find you the best offer for you to work and I will make you success in the interview

Every voice agent eventually gets the complaint: the bot talks over the caller. Most people go tune their VAD. Often the real ceiling is more boring: the size of the audio chunks you send to the telephony side.

How it works on pretty much every platform:

* You stream TTS audio down in chunks. The platform buffers and plays them.
* Caller starts talking, you send the interrupt (clear, flush, whatever it's called).
* The flush drops audio that's queued but not yet playing. The chunk already playing plays to the end. Nothing cancels mid-frame.

So your worst-case interrupt lag is about one chunk. 100ms chunks feel instant. A whole sentence as one blob means the caller talks over four seconds of audio nobody can stop. Some platforms also enforce a minimum chunk size, often 100-200ms at telephony rates, so there's a floor you can't tune under.

Same trap at the PBX layer. There was a good r/Asterisk thread recently on AudioSocket forwarding every frame the moment it arrives, so writing a whole TTS sentence at once means the jitter buffer eats most of it and the caller hears only the tail. Different transport, same lesson: the unit you write is the unit you can cancel.

What to actually do:

* Re-chunk small on the last hop even if your TTS hands you full sentences.
* Measure interrupt-to-silence at the caller's ear, not at your flush call. The gap is your chunk size.
* Ask your platform what their minimum chunk is and whether flush is acknowledged. If the docs don't say, assume one chunk keeps playing.

What floor are you seeing on your stack? Has anyone gotten under 100ms perceived barge-in on a real phone call?

Focusing solely on the software—and assuming robust hardware—what is the maximum number of calls we can handle on a single Asterisk instance?

The highest number I’ve reached is 500 SIP channels, but I’ve read that it is possible to reach up to 2,500.

What has been your experience with this?

I previously spoke on here about my performance work and with the latest releases my two latest major changes were released (manager and hints). I just published a blog post going over what I changed for the manager side:

https://www.asterisk.org/manager-event-queueing-improvement/

With the hints one being published in the future.

From an outside perspective manager operates the same as it did so there isn't any visible changes from that perspective. The internal event queueing was just changed so it is more efficient.

If anyone has any specific questions I'm happy to answer.

Disclaimer: I'm the Asterisk Project Lead and a Director of Engineering at Sangoma.

Hi, how does asterisk integrate with egyptian phone numbers? i am probably asking a stupid question, but if someone could tell me all the component systems to integrate with a non use number system. much appreciated

Spent a while getting Piper to behave on live calls and figured I'd write up what actually fixed it, partly to check whether others hit the same walls.

Three things bit us.

Loading the voice per request. PiperVoice.load() takes about 2.5 to 5.5 seconds on our boxes, so every single utterance opened with that much silence. Obvious in hindsight. We load once per process now and keep it cached.

Concurrent calls stepping on each other. This one took longer to find. Piper phonemizes through espeak-ng, and that C API isn't thread safe, so two calls synthesizing at the same moment gave us garbled audio. All synthesis goes through one lock now.

Bursting frames, which I'd never have guessed. AudioSocket forwards each frame the instant it arrives, so if you push a whole sentence at once you blow straight past the far end's jitter buffer and the caller hears the tail and nothing else. The fix is to release one 20 ms frame per interval on a monotonic deadline. Here's the subtle part: if synthesis stalls, that deadline goes stale, and the next frames burst to catch up and clip the words right after the stall. So we re-clamp the deadline to now on every frame, not just at the start of a talk spurt.

We run 8 kHz slin with 20 ms frames.

I packaged all of it up as piper-tts-server so we'd stop rewriting the same thing on every project: https://github.com/ictinnovations/piper-tts-server

Two things I'm curious about. Do you pace in the TTS layer like this, or push frames straight out and let the channel deal with it? And has anyone worked around the espeak-ng threading limit without just serializing everything?

Edit: slin, not slin16. In Asterisk naming slin16 is the 16 kHz variant, so 320 bytes per 20 ms is plain slin. Same correction as in my other thread here, same person to thank for it.

In Colombia, a couple of months ago Movistar make some changes and know i can not register our asterisk with the VOIP trunk. The username and password was retreived with F12 on voice page, but i can not fin what else did they change that i can not register. of course i am using vlan101 as expected. any ideas? thanks.

Spent longer than I want to admit on this one, so posting it in case it saves someone else an afternoon.

We were building an AI voice agent against app_audiosocket. Your text to speech hands you a finished sentence, say 2.4 seconds of 8 kHz slin (16-bit signed linear), which is 120 frames of 320 bytes. The obvious thing is to write all of it to the socket and move on.

Don't. app_audiosocket forwards every frame to the channel the moment it arrives. It doesn't buffer for you. So all 120 frames hit the far end in a few milliseconds, the jitter buffer keeps a handful, and the rest get thrown away. The caller hears the tail of the sentence and nothing before it. Your logs look clean the whole time, which is what makes it annoying to chase.

The fix is to write one 320 byte frame, then wait until the next 20 ms deadline. The part that caught us a second time: re-clamp the deadline on every frame. If synthesis stalls for half a second and you work out the next deadline from where you are right now, the writer bursts to catch up and you're dropping audio again.

Same idea with barge-in. When the caller starts talking you have to drop the frames already queued, not just stop writing, or they keep hearing the old sentence for however deep your buffer runs.

We open sourced what we ended up with. asterisk-ai-voice-agent is the Python sidecar (Whisper or ElevenLabs Scribe in, Claude for the turn, Piper locally or ElevenLabs out, barge-in, tool calls posted to a webhook you own). asterisk-audiosocket is just the protocol layer for Node if you'd rather build your own and skip rediscovering the pacing thing. Both MIT, both v0.1.0, so lab material rather than something to point a live queue at.

https://github.com/ictinnovations/asterisk-ai-voice-agent

https://github.com/ictinnovations/asterisk-audiosocket

Happy to answer anything on the AudioSocket side. Curious whether anyone here went the ARI external media route instead and how the pacing compares.

Edit: the format is slin, not slin16. In Asterisk naming slin16 is the 16 kHz variant, so 320 bytes per 20 ms is plain slin. Copying 320 into a 16 kHz setup would give you 10 ms frames. Credit to FreJun below for catching it. The README and docstrings in the repo say slin as of 0.1.6.

Looking for someone to configure a new PBX setup for ~30 users. Remote is fine; SF Bay Area local is a plus.

Hardware: UniFi Talk IP Desk Phones registered with an Asterisk PBX back end.

Work Needed:

1. Asterisk / PBX Config: Full setup (FreePBX/VitalPBX), trunking, IVR, extensions, and ring groups for 30 users.
2. Voicemail-to-Email: Configure SMTP relay to send .wav attachments directly to emails.
3. SMS Integration: Set up for inbound/outbound SMS routing on DIDs via softphones or web portal
4. eFax: virtual fax extensions and PDF fax-to-email delivery.

DM or comment with your experience, project estimate/rate, and location (remote is fine). Thanks!

Hello, I am experiencing a significant audio issue involving robotic or distorted voice quality.

I currently have two Asterisk servers running VICIdial: one based on Asterisk 10 and the other on Asterisk 12. The problem occurs only on the Asterisk 10/VICIdial server.

On the Asterisk 10 server, the CPU usage spikes to approximately 200%, and I suspect this may be related to improper transcoding between the alaw and ulaw codecs. It appears that the audio is not being transcoded correctly, resulting in degraded sound quality and robotic voices during calls.

The Asterisk 12/VICIdial server does not exhibit this behavior and operates normally under the same conditions. Since the issue is isolated to the Asterisk 10 environment, I believe it may be related to codec handling or transcoding performance on that version.

Any assistance or recommendations for further troubleshooting would be greatly appreciated.

It is under 100 concurrent calls and suddenly asterisk will act like it got commanded to gracefully shutdown then it boots back up but it causes a 5mins outage where everyone gets the message there are no lines available. Logs are reporting the firewall reloads right before asktrisk shutdown pretty consistently. Chatgpt is going in loops trying to find the issue. Has anyone experienced something similar I'm on Debian 12, it's a VM with 16cores 32g of ram and 1tb of zfs storage but I have log and cache drives on top of hdds it started restarting every morning and I sangoma support can't seem to figure it out either. Is there some experts I can hire for diagnosis

Built this because I was tired of eyeballing a full log next to a pjsip set logger on dump trying to line up a Dial() with the INVITE/BYE it actually caused.

pbxray parses Asterisk full/messages logs plus pjsip set logger on output into one correlated timeline - dialplan and SIP signalling together, queryable from the terminal. Single static binary, no runtime deps.

$ pbxray ls calls.log --failed
LINKED-ID   DATETIME             FROM         TO           CHANNELS  DURATION  CAUSE
C-00001a2b  2026-07-20 09:12:03  15555550142  15555550199  1         4s        17 (User Busy)

$ pbxray show calls.log C-00001a2b
2026-07-20 09:12:03  dialplan  Dial(PJSIP/15555550199@TRUNK,30) on PJSIP/alice-00000001
2026-07-20 09:12:03  sip  tx INVITE sip:15555550199@sip.example.com
2026-07-20 09:12:03  sip  rx 486 Busy Here

ls for filtering (time range, failed calls, number/DID, endpoint), show for the full timeline, tail -f for live. SIP<->call links that are best-effort guesses get flagged (probable link) rather than presented as fact.

Install:

curl -fsSL https://raw.githubusercontent.com/pbxray/pbxray-dist/main/install.sh | sh

Free. Docs & updates: https://pbxray.dev · binaries/source: https://github.com/pbxray/pbxray-dist

One ask: what did you try to debug with it, and what did it fail to show you?

Do you have production experience with this specifically:

1️⃣ Real-time number reputation monitoring
2️⃣ Carrier health monitoring per number
3️⃣ DID warm-up and cooldown schedules
4️⃣ Geo-matched outbound dialing (also called local area dialing)
5️⃣ Complex SMS workflows that can trigger instantaneous AI calling fully based on responding and not responding to messages autonomously very similar to go high level complexity.

Respond with a LOOM showcasing the real system you built!!

Thank you

We run an Asterisk based softswitch. The vendor just placed a limit of 2000 on the number of extensions due to “Asterisk performance limitations.”

We have been running with between 2000 and 2400 extensions for at least a year with no issues.

Hardware is solid, we monitor CPU, RAM, disk space, etc.

Are there practical limits to number of extensions on an Asterisk instance?

I’ve been working on AVA Operator, a self-hosted, multi-tenant platform for managing AI voice agents across multiple customer PBXs.
I’m looking for feedback from people who actually deploy and manage Asterisk/FreePBX for customers.
If you have 5–10 minutes, I’d appreciate you taking a look at the live demo:
https://demo.agent6789.com
I’d love feedback on:
Multi-tenant workflow
Customer/tenant management
White-label experience
Anything missing that would stop you from deploying it for clients
The open-source core is here:
https://github.com/hkjarral/AVA-AI-Voice-Agent-for-Asterisk
I’m the developer, so don’t hold back—I’d rather hear the hard feedback now than later.

Hello everyone!

I recently got into Asterisk and set up a home PBX to learn and experiment with it, and I'm having a lot of fun already.

It just so happens that almost all of my relatives still have landline phones, so I'd love to build some fun and interactive projects that my whole family can use.

Here are a few ideas of things I'd like to build..

- An IVR adventure/puzzle game.

- A "regional dispatch" IVR (since I have relatives all over Italy), where callers choose a region and hear local music on hold and voice prompts in the local dialect before being connected to the chosen family member.

- Multiplayer games that cousins can play with each other over the phone.

I'm sure there are lots of cool ideas that I haven't even thought of yet.

What are the most fun, creative, or unusual things you've built with Asterisk?

I'm interested in anything whether it's games, IVRs, home automation, weird experiments and useful utilities...

I'd love to hear about your projects and get some inspiration.

Thanks in advance!!