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I am switching from FreePBX to FusionPBX and we have a ton of Grandstream HT286 among our end users that still work perfectly fine with Asterisk. However, I can not get them to register with Fusionpbx. I entering the SIP credentials manually and I can see the SIP auth challenge in the log but...

Does anyone know how to provision a SPA 2102 from Fusionpbx? I followed the instructions and added the profile rule below and rebooted the SPA but it did not provision. [--uid provision-user --pwd provision-password]http://mydomain.com/app/provision/?mac=$MA Typically I use the syntax below to...

I am trying to provision a Grandstream HT/701. After setting everything up on the fusionpbx side, I opened the Grandstream GUI and entered the config server path, http user name and password. If I set provisioning to HTTP, it works. HTTPS does not work. We use Let's Encrypt for SSL. Is there...

I have a ticket open for this so if anyone experiences a similar issue where a fax is received by FusionPBX but it does not trigger the email, PLEASE open a ticket . I suspect they are waiting for other people to report the same issue to fix this.

What about automating the Letsencrypt renewal with webmin? Has anyone tried that?

I installed Let's Encrypt with the dehydrated script and was under the impression that it would somehow auto-renew but I always have to renew manually. Is there a way to set it up so that it renews the SSL certificate automatically without any intervention? When I do it manually, I always have...

Maybe I misunderstood the issue but if it works the way you want when you use Zoiper and you confirmed that everything is handled by fusionpbx, then why not increase the log level and debug level to 7 and study the log of the call to see which transfer function Zoiper is using in fusionpbx?

I found the issue. Apparently Anveo has a really low monthly spending limit by default and I exceeded it. Thanks guys.

Unfortunately, that did not do it. Does anyone have this working? Anyone using Anveo as provider on FusionPBX?

Actually this would be really easy to check. Simply look at your fusionpbx log during a Zoiper transfer from A to B. If you see nothing then the transfer is done via Zoiper PBX.

Anyone know how to setup an Anveo Gateway? The host is sbc.anveo.com Do I have to allow additional signaling IPs in the ACL list? In FreePBX everything works great but same settings in Fusionpbx fail. I keep getting CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]

Yes. That's what I did but I was hoping there is a better way. I created an internal outbound route (order 50) with regex ^1(\d{10})$ that matches my 11-digit Destinations. Now every outbound call checks the internal route first and transfers the call internally if there is a match. It works...

^0\d{6,20}$ catches overseas calls starting with zero but I think I get the idea. ^0\d{6,20}$|^1\d{10}$|(^911$|^933$) regex in a 'call_direction-outbound' rule should catch USA, overseas and 911 calls.

${effective_caller_id_name}, ${effective_caller_id_number} did not work. I'll try ${sip_from_display}. Thanks.

Normally the scenario you are describing is only possible when A, B, and C are all on the same PBX system. If B and C are on POTS lines then A can not release the line after establishing a 3-way conference. This would disconnect B and C. I am still learning fusionpbx but if they have an...

So you delete all 'call_direction-outbound' rules and you keep only one with regex ^0\d{6,20}$ to catch all routes? If that works for proper CDR then it's much easier than maintaining multiple 'call_direction-outbound' rules.

Thanks but that did not work. The 'user_exists' condition probably looks only at extensions, not inbound routes.

The above scenario describes a 3-way conference call. Your agent calls the first party and puts them on hold. Then he calls the second party and connects everyone or connects the 2 parties and exits the call but there is no call from the first party to the second party in this scenario.

My PBX has 11-digit Destinations (Inbound Routes) and 10-digit Extensions. If extension 1000 dials extension 5141234567 then the call is processed as an internal transfer. However, if extension 1000 dials Destination 15141234567 then the call is sent to an Outbound route and processed via an...

Not sure if this is a problem, but I noticed a couple of issues with the new "call_direction-outbound rule" that is now created for each new "outbound route" 1. Initially the Regex in the "call_direction-outbound rule" is the same as the Regex in the matching "outbound route". However, if I...