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That's really too bad. It would be great to switch back from XML to GUI (dialplan_detail records) for inbound/outbound routes and dialplan manager records. Seems like this is something that should be added in a future date. Not sure how it works in this forum but can we start a bounty to add...

In Dialplan manager I can view a call destination (for example call_screen) in the Gui and also in XML code. However, when I make changes in the XML code and save them, those changes are not saved to the dialplan GUI. Is this supposed to work? It would make sense that FusionPBX allows switching...

Can anyone help? I changed the code as described above and it prompts for the digit 1 when I enable call screening but when I press the 1 it is ignored and the call disconnects. There must be something off on the last line of the code.

Apparently it's minimum $100 per month

Thanks for the suggestion, it looks interesting. If I understand correctly, the IVOZ application is a Multitenant PBX. It's an integrated billing solution that's built around Asterisk 13 and PJSIP but I want to move away from Asterisk to Freeswitch so this won't work for me. My goal is to use...

Definitely good

Yes, I sort of feel bad about it but I hadn't gotten any response yet and thought hijacking my thread would distract readers from the actual message in my post. However, no matter what the developer did in the past, if this Billing module works as advertised (and feedback so far seems to say it...

I contacted the developer via messenger and his website is back up. Apparently a DNS issue on the server. I will also try to install it later this week and go for the $300 CAD license. It's not a big investment. It's cheaper in the long term and brings us one step closer to going open source.

It's not free. There is a recurring monthly fee of $49 (I think) or a one-time fee of $300 for the license. I haven't tried it yet which I why I asked for feedback here.

It's not abandoned. Just takes a long time to load. Try this link. https://okay.network/communications/freeswitch-fusionpbx-goodies/billing-for-fusionpbx-with-freeswitch/billing-for-fusionpbx-installation.html

I don't know if it's really necessary to write a whole new billing module when there is already one available that can potentially do the job. That's what this thread is about. To get some feedback from users on this forum who have used the Billing module from Okay Network. There must be...

Can you check your code if it's compatible with version 5.0? It just repeats "To continue press 1" and disconnects even if 1 is pressed.

Coming from the FreePBX / Asterisk community, I am a bit disappointed about the level of engagement and help on this forum. There must be some users in this forum who have used the Okay Network LCR and billing system for fusionpbx but when I search this forum there is almost no mention about...

Please don’t hijack my thread with your advertisement. I would appreciate an answer from actual users if the above mentioned billing system. I just find it weird that this forum does not mention it at all.

Is anyone here using the fusionpbx billing system by Okay Network? I'd like to get some feedback on this application, please. It sounds like a better option than trunking every outbound call to ASTPP. Many thanks in advance for your help.

GS Wave Lite does not need push notifications for incoming calls. It works fine running in the background with TCP protocol.

GS Wave does not need push notifications for incoming calls. It works fine running in the background with TCP protocol.

Thanks but that's not it. The codecs are loaded in Modules. However, I got it to work in the end. After changing the codecs in Advanced->variables, I had to go to Advanced->sip profiles and SAVE the internal or external profile. Then go to Status-sip status and restart the internal and external...

I am new to fusionpbx and could not find the info in the forum. I am using Acrobits to register an extension and can make and receive calls with ulaw and G722 codec. However when I try either Opus or GSM codecs the call fails. The codecs are loaded in MODULES and show codec in fs_cli lists the...

Actually, we have no issue in IOS with GS wave incoming calls if we use the TCP protocol. The app runs in the background and receives all incoming calls. The real problem with GS Wave, is the 2-3 seconds audio delay for all incoming and outgoing calls. We even opened a ticket with Grandstream...