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Your fist one should work but you will need to set the "inline" flag on the set action otherwise the max_calls variable will not be available for the limit action. You may also want to use a more simple name for the realm without the (, |, and) characters.

Have a look at https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/3375201/ When you say "limit the number of incoming and outgoing channels per domain", I assume you mean inbound and outbound calls via gateways not simply channels connecting two extensions. If...

Unless it's changed in newer versions of FusionPBX, you can just select any of the ${sip_ choices and then click on it to edit it to whatever you want. Screen shot below:

Yes, so the lower capture confirms it, there is fragmentation going on. The choices are 1. Switch to TCP (bigger system overhead) or 2. Reduce your UDP packet size. One of the easiest way of reducing the UDP packet size is to limit the number of codecs you offer, thus reducing the size of the...

Maybe start by looking at your UDP packet sizes. If the packet size is exceeding the MTU, then it will be fragmented, TCP is unaffected by this. I have seen it many times when an initial INVITE message is OK, but then the proxy responds with w WWW-Authenticate, the client sends the INVITE...

@alan If I understand your original question correctly, you are asking how to set the caller_id in the dialplan, if so, then just set effective_caller_id_number=+1xxxx etc. If you are trying to make a call from India, look like it originated in the USA, your call may get blocked or have the CLI...

I apologise for butting in here, I noticed that in your original post you mention cross talk, it was common in the analogue telephony days, but now few people take reports of it seriously with VoIP, however there are ways it can happen. If nothing had changed in your Fusion/Freeswitch...

Thank you for that, and I agree about the spam problem. I sometimes find 2FA difficult, because I work in places that don't allow mobile phones, and I don't always have an email account setup on the laptop I may be using at the time. I do generally have a soft phone though, so I can call...

I was forced to enable two step verification this morning when logging in here. I see there is an option to disable it, but if I do , will it just force me to enable it again?

I wanted to use python in preference to Lua whilst developing DjangoPBX. But I found the interface buggy, I could never track it down but there seemed to be some memory corruption between successive calls to Python within the FreeSWITCH module. I did submit some pull requests to FreeSWITCH...

We bought the ones we use many years ago, the vendors are still going, link below: https://www.westany.com/united_kingdom_voice_prompts/

I have no plans currently. But there is no reason why it can't be done. You can configure the bindings in the SIP profiles and adjust the firewall rules accordingly, the only caveat is that there is no hook currently for a successful webrtc registration to place a whitelist rule into the...

@AGIDI Thankyou we have run on XCP-Ng for years with no problems, but are seriously considering moving to Proxmox, I just haven't managed to work through the find detail yet. I wondered if an LXC Container would be better for FreeSWITCH than a VM.

Does anyone have a view on virtual machine vs LXC containers?

In your Dialplan screen: Remove the data=" and the expression=" text. Remove the "/> from the end of each line. Finally the set actions should come before the Transfer action. PS: Please post questions only once.

Hello One & All, I know many people/organisations choose to use Proxmox for their virtualisation. I find Proxmox interesting because it offers the choice of running an LXC container as a lightweight alternative to fully virtualised KVM machines. If you use Proxmox, how you virtualise your...

Hi @manikanta , I'm sorry I do not know the answer. Are you absolutely sure it is WebRTC related? The 403, would suggest that FreeSWITCH received the response to the WWW-Authenticate OK, but either (username or password was incorrect) or FreeSWITCH was unable to retrieve the directory...

In your inbound route put the caller ID into a hash store: Then create a dialplan entry called last_caller with a destination number of 1471, see screenshot below:

Thank you for the information, I have not seen MirtaPBX and I will drop you a PM. I haven't used Asterisk for a long time, I have compiled it from source many times in the distant past and also played with distributions like Trixbox. When we ran just purely a SIP trunking platform I used...

Hi Alan, you are very welcome.