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This system only has 2 DID's on it. The server is hosted on VULR. I don't have SSL setup or anything would that cause this? I've never had this issue before.

Does anyone know why this error happens? How can I fix this? Could this cause incoming calls to not work? I don't get dial tone and the call just hangs up.

I also noticed if I have an extension registered in two places, it only rings on one of the phones. Right now I have a Yealink Desk phone and Zoiper (Paid service) and only the desk phone rings. Logs aren't really showing me anything however I am seeing this with the phone that doesnt ring and...

For one of our DID's on the same tenant incoming calls do not work. When you call the number you don't hear anything and it immediately just hangs up. I tried registering the extension on another phone and its the same issue. Anyone know what could cause this? I have another DID in the same...

I knew it was something simple. Thanks! @Bifur

Hi, Having trouble making inbound calls from my cellphone to my DID. Making calls from my DID to my cellphone works great along with CallerID etc. The other way is not working and I cannot seem to figure out why. Does anyone know why it could be? I have a SIP trunk through IXICA if that...

Is there a way to have a toll free number forwarded to a local DID but not have the Toll Free leg still active? Right now if I have a Toll Free number forward to a local DID within the same system I essentially get billed for the Toll Free AND the Local as they both stay active during the call...

anyone have any ideas?

Pretty simple question I hope... When I want to add a SSL Certificate to my pbx for the FusionPBX portal, will this impact any existing setups that I have? I am going to follow this link https://docs.fusionpbx.com/en/latest/getting_started/lets_encrypt.html . Is this the correct link to follow...

I see the below: error.log 2020/01/28 06:26:02 [error] 783#783: *502029 upstream timed out (110: Connection timed out) while reading response header from upstream, client: 1.1.1.1, server: fusionpbx, request: "GET /app/fax/fax_emails.php HTTP/1.1", upstream...

Any other things I can check? Doesn't appear to be a fail2ban issue when I attempt and also logs don't show anything either. My problem seems to be the fact that /app/fax/fax_emails.php']https://[MYSERVER]/app/fax/fax_emails.php cannot be reached. I think if that could be sorted, this may work.

I am able to get to my provisioning server just fine though (/app/fax/fax_emails.php']https://[MYSERVER]/app/provision/). If the WAN IP was in fail2ban i believe that would not work correct?

anyone have any ideas on this?

I have searched the forum and have found some useful info in regards to sending email to fax by setting up the advanced section under the FAX server however I am showing a gateway timeout when i visit https://[myserver]/app/fax/fax_emails.php in my browser. My understanding is if i visit this...

Where does one edit the flush cache permissions for a user. I am going to Status and then Sip Status and then at the top looking for where to set this. I have made changes under menu manager and allowed the status one and also the sip status one. I also made changes under group manager and under...

Thanks for the information. Right now i have the inbound route as 11234567891. Our users always typically always dial either the extension, 4 digits in our case which of course is working internally or they dial 1234567891 I don't want to make changes that would mess anything up. How can I set...

Should FusionPBX know that a call is internal by default? When we dial our 4 digit extension obviously that is internal and all is well. What about if you dial their 10 or 11 digit number? (234-456-8900 or 1-234-456-8900) Should Fusion know to treat that as internal or does a dial plan need to...

Thanks for the reply. What happens if we were actually open on holidays but for a shorter period? Does that mean I have a separate time condition for all holidays and the custom times and have the routing go to that first? for example Time condtion 1: 6am to 11am monday to friday, weekends...

This has been sorted out. Just ended up editing template.php. When i go ahead and update fusion, does that mean any changes made to template.php will get changed back to defaults?

I had this issue not long ago and it was due to video codec's being sent which caused about a 4 second cut off at the beginning of the voicemail being recorded along with when recording the greeting as well. I resolved this by adding the below to my vm_user under dialplan manager. This needs to...