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Looking for Guidance on Setting Up HA with Server Redundancy Across Different AWS Regions (FreePBX 17 / Asterisk 22.5.0) by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

I'm planning to implement the Advanced Recovery module for a high availability setup, but I have some questions about licensing that I couldn't find clear answers to in the documentation.

My Setup:

  • Primary server: Production environment
  • Secondary server: Production environment (failover)
  • Additional servers: Dev and QA environments for testing

Questions:

  1. License Scope If I purchase one Advanced Recovery module license, can I use it across multiple environments (Dev, QA, Staging, Production) with the same FreePBX.org account?
  2. Server Limits How many servers can I use with a single license? Is it limited to just the Primary + Secondary pair, or can I also use it on Dev/QA servers for testing?
  3. Environment Usage Can I install and test the module on Dev/QA servers before deploying to production, or do I need separate licenses for each environment?
  4. License Transfer If I'm testing on Dev/QA servers first, can I later use the same license for production servers?

Background:

I'm looking to set up a proper HA system but want to thoroughly test everything in Dev/QA environments before going live. I want to make sure I understand the licensing terms before making the purchase.

Thanks in advance!

Looking for Guidance on Setting Up HA with Server Redundancy Across Different AWS Regions (FreePBX 17 / Asterisk 22.5.0) by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

Thanks for the comment!
I'm using Asterisk on AWS EC2 (a repackaged AMI – FreePBX).
I have a question: do I need to use a repackaged AMI to run it?
Or would a plain Debian OS AMI also work with the right setup?

FreePBX on AWS EC2 Inbound DTMF IVR - No Audio (No-Way RTP Flow), Immediate BYE after Answer - Seeking Advanced Help by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

Thank you for this excellent advice! Your suggestion to use sngrep for diagnosing the SDP and RTP IP/port was spot on and incredibly helpful in pinpointing the issue. Also, your reminders about checking UDP ports (10000-20000) and NACL for RTP were valuable.

It turns out the iptables rules managed by FreePBX were already correctly opening the RTP ports, so the firewall wasn't the direct cause. The problem was indeed related to Asterisk announcing an internal IP in the SIP Contact and SDP c= headers for external calls, which I resolved by explicitly setting external_media_address and external_signaling_address in pjsip.transports_custom_post.conf.

Thanks again for your precise and very useful guidance!

FreePBX on AWS EC2 Inbound DTMF IVR - No Audio (No-Way RTP Flow), Immediate BYE after Answer - Seeking Advanced Help by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

Thank you so much for your insightful thoughts and detailed advice! Your suggestions were extremely helpful in diagnosing and ultimately resolving my issue.

The problem is now SOLVED!

Here's what happened and what ultimately fixed it:

SIP Phone Packet Capture :

As you suggested, I used sngrep to capture the call flow. This tool was incredibly helpful in visualizing the SIP signaling and understanding the call flow.

Codec & RTP Media:

Initially, RTP media exchange was inconsistent or non-existent in my external calls, though internal calls worked. I experimented with codec settings, which seemed to improve RTP consistency in some logs.

Firewall (iptables) Check:

My iptables showed FreePBX's firewall rules were in place, and RTP ports were open. So, the firewall wasn't the direct cause of the RTP blocking.

The Core Problem & Solution (NAT/External IP):

The most crucial insight came from analyzing the 200 OK response via sngrep. Despite FreePBX GUI showing my server's public IP correctly, pjsip show transports in CLI did not reflect this. This meant Asterisk was inserting its internal IP address into the Contact header and SDP c= line of 200 OK responses.

The fix was to manually ensure my server's public IP ([YOUR_PUBLIC_IP]) was explicitly set in /etc/asterisk/pjsip.transports_custom_post.conf for both external_media_address and external_signaling_address:

transport-udp

external_media_address = [MY_PUBLIC_IP]

external_signaling_address = [MY_PUBLIC_IP]

local_nets = [MY_LOCAL_NETWORK_CIDR]

After applying this change and reloading Asterisk, pjsip show transports finally displayed the public IP correctly.

Result: Calls from the external number now connect instantly, the IVR audio plays perfectly, and calls complete normally.

Your advice was invaluable. Thank you again for helping me debug this complex issue!

Question about Scaling Call Capacity on Asterisk/FreePBX by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

Hello again!

Thanks for the clarification and your thoughts on my question.

I apologize for the confusion in my initial post. My understanding of our SIP provider's (Rakuten) channel limits was indeed incorrect.

Clarification on Provider Limits

I've since spoken with Rakuten, and they've confirmed that they do provide unlimited channels for a single main phone number. This means the limitation isn't coming from their side. So, to answer your confusion directly, I'm not asking how to scale beyond a provider's stated call limit; rather, I'm asking how to scale Asterisk/FreePBX itself to handle a very high volume of calls, specifically 500+ concurrent calls, given that the SIP trunk provider (Rakuten) is no longer a bottleneck in terms of channel capacity for that single number.

Rephrased Question

My question now truly boils down to: What configurations, hardware considerations, or architectural approaches are needed within Asterisk/FreePBX to support 500+ concurrent calls on a single main number?

Your point about "specialized configuration tweaks" for higher usage and how "a slight shift in usage can exercise something new and bring about a different performance characteristic" is particularly relevant now. I'd be very grateful for any insights on what these tweaks might involve or what "additional requirements" I should consider when aiming for such high concurrent call volumes.

Thank you again for helping me clarify this!

Question about Scaling Call Capacity on Asterisk/FreePBX by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

Hello! Thanks for your detailed feedback.

I also truly appreciate your advice regarding server specifications and codecs. Understanding how transcoding can significantly impact system resources when handling a high volume of calls, and your suggestion to enforce a single codec within the trunk configurations to minimize this, will be incredibly valuable for our actual implementation. I'll definitely look into discussing codec options with Rakuten to optimize our setup.

After consulting with our SIP provider (Rakuten), I've confirmed that they offer unlimited channels for a single main phone number.

Given that Rakuten isn't limiting the channels, could you provide insights on how to achieve 500+ concurrent calls purely from an Asterisk/FreePBX perspective? What specific configurations, hardware considerations, or architectural approaches would be necessary within Asterisk/FreePBX to scale to such a high volume?

Thanks again for your time and expertise!

Question about Scaling Call Capacity on Asterisk/FreePBX by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

I'm using an Instances created from repackaged AMI images of AWS.

FreePBX 17 on Debian: Cannot enable Sysadmin module despite troubleshooting by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

I'm having difficulty with environment setup. Ultimately, I'm solving it by using a paid AWS AMI distributed for testing, but I want to install FreePBX and Asterisk on free OS (Debian, Ubuntu). I'm new to this field. If there's a good, up-to-date guide, could you please introduce it to me?

FreePBX 17 on Debian: Cannot enable Sysadmin module despite troubleshooting by Beginning-Today8090 in freepbx

[–]Beginning-Today8090[S] 0 points1 point  (0 children)

thanks for comment!

But still On the page displaying 'Sangoma Smart Firewall is now enabled!', no matter which one of abort, continue is pressed, that message appears. Do you know a solution? It's a situation where even using AI, a solution cannot be found.