Recent content by businesstelecom

Hello everyone, After working with various billing platforms over the years — including ASTPP, WHMCS modules, and several custom setups — we found that none of them fully met the requirements for a stable, lightweight, and customizable retail SIP trunk billing environment. Because of this...

Hello everyone, After working with various billing platforms over the years — including ASTPP, WHMCS modules, and several custom setups — we found that none of them fully met the requirements for a stable, lightweight, and customizable retail SIP trunk billing environment. Because of this, we...

Hi, I work with multiple FreeSWITCH servers and I'm trying to use AI tools to help generate dialplan XML, gateway configs, routing logic, and troubleshoot SIP issues. I’ve tried ChatGPT, Claude, Gemini, and GitHub Copilot — but none of them produce 100% accurate FreeSWITCH configuration, even...

I’m building a VoIP billing system using FreeSWITCH and the XML CURL API. So far, SIP registration, SIP profiles, and gateways are all working dynamically from the database. Now I’m working on the dialplan part — and I’m a bit stuck. I want to know the correct XML format that FreeSWITCH...

thank you very much.. is there any guide or something somewhere? or can you please help me out ? i just need to understand how the things connected with xml api thing.. thank you

Hello everyone, I’ve been working with FreeSWITCH for a while and I’m trying to build a simple SIP call server that can handle SIP trunk routing, DID management, and billing. I understand how FreeSWITCH works with XML dialplans and mod_xml_curl, and I already have a working XML API that...

Hello Thanks very much to the admin who made this wonderful group. i am facing again a very strange problem with the fusionpbx basically i have 1 ASTPP server. in that server i have one SIP account and 5 DID number forwarded to that SINGLE SIP Account. so if i login that sip account directly...

thanks for your reply. its basically they have 3 quality of routes. 1. premium which is 4000 prefix 2. gold which is 5000 3. standard 6000 so based on the prefix the routing works. but i only see this in the yealink phones. not any other phones like grandstream thanks

hello sir can you please help me another thing. basically my voip server i have added in the termination side prefix for my gateway provider. as they told me i need to send calls with prefix 4000 not direct international format so i send them like 4000+353+mobile number but in my yealink phone...

thanks dear. i have checked those file already.. but its almost same description for everything. i wanted to know more like the real reason thanks

Hello everyone can someone guide me how this fail2ban works? i mean when fail2ban ban an IP will this IP send to IPtables list for ban? and how do i see that iptables ban list? i mean any command for it? i can see there are two file called jail.conf and jail.local in both files there is jail...

super.. i just add this to my sip profile.. and its working straight away.. big thanks and salute to you sir.

hello hfoster. you are a genius. yes i just go the yealink phone page and enable the rport and its worked. but can i not just change something in my server so it will be work without the rPort enable in yealink phone. as i have tested with other voip accounts in that yealink phone without open...

Hello I am finding a very strange issue with configuring a Yealink phone. Fusionpbx extension connecting every sip application like zoiper microsip. But it's not connecting to the Yealink phones. I have tried that phone with other company sip account it's registering. Can someone face this...

Hello everyone Hope everyone doing well. I installed fusionpbx on a AWS ec2 server with latest version on debian 11. going to put one company pbx with few groups with few extensions. But i am little scare with the security as voip is too risky project. i have couple of questions. may be you...