Search results

Ok, after much trial and error aka learning the hard way I figured this out. So basically ASTPP comes with only one profile out of the box. I was trying to do everything through this one profile and as a result the SIP headers would get mucked up one way or another and I'd get an ACK timeout...

We took a run at this a couple of months ago and ended up just going to a proprietary SBC. We're still experimenting but basically we're probably going to put an AnyNode SBC in between the Microsoft stuff and our Freeswitch stuff. After many hours of Googling I only heard the odd whisper of...

I emailed you the sngrep pcap if thats alright. basically the call ends because it's not getting an ACK back (408) There's a lot of NATing going on so its not surprising.

Fusion PBX = Version 1.6.20 64bit ASTPP = Version 1.6.19 git 7a77e0b 2017-07-13 12:01:45Z 64bit

Hi Guys, Disclaimer: Yes I know my test platform is weird, with that said, Here is my setup Public IP A and B go to pfsense firewall. Fusion PBX box and ASTPP box are behind the firewall on a private network. Public IP A is nated to the fusion PBX box on all the appropriate ports for SIP...

Coming in late to this. As mentioned previously there are 2 ways to go with Cisco phones: 1) Cisco SPA, which is derived from sipura phones making them fairly generic and easy to work with. 2) Cisco everything else (79XX, 88XX, etc) which are built to work with call mangler. Cisco SPA is end...

I have one server running Centos and I wanted to run the the letsencrypt script to setup ssl and TLS. It turns out you only have to make three edits 1st the sed statements simply need to point to fusionpbx.conf instead of just fusionpbx: sed "s@ssl_certificate...

I know its not officially supported but I have a client that wants to use us for outbound calling from teams. Of course there's a supported SBC list but it looks like its just TLS and SIP over TCP. Anyway before I try to re-invent the wheel I'm wondering if anyone else has taken a crack at this?

You could use a device like an Algo 8180 to make the announcement. I'm not sure how Fusion PBX handles paging though.

Sorted. This was the script as copied from documentation: https://docs.fusionpbx.com/en/latest/getting_started/backup.html #!/bin/sh now=$(date +%Y-%m-%d) echo "Server Backup" export PGPASSWORD="zzzzzzzz" mkdir -p /var/backups/fusionpbx/postgresql #delete postgres logs older than 7 days find...

Great script, thank you! It worked so there must be a problem in my backup script.

Hi, I'm trying to back up the database using the backup script in the documentation. I've plugged in the password from the /etc/fusionpbx/config.php file but when I run it from the shell as root I get: connection to database "fusionpbx" failed: FATAL: Peer authentication failed for user...

Thanks it was a carrier issue. Case closed.

I have a problem prefixing an out bound route. Its a pretty basic route to accept a ten digit number and then prefix it with a one before sending it to the gateway. I need to prefix it with a 1 so that it gets properly rated on an ASTPP box (the gateway the call is being sent to). Anyway this...

Ok, I fixed it and of course it was something stupid. These are test machines and because of that I'm using the stock iptables rules they both came with during setup. I guess I assumed because they're both Freeswitch projects that the products would've been operating on the same RTP range...

The only thing Nating to the Fusion PBX is the phone that registers from the public network. The link between ASTPP and Fusion is on the local subnet. I'm going to simplify and register a phone to ASTPP on the local network and see if I have the same problems.

Ok, so I've probably made this install more complicated than it needs to be but here is my issue: Calls set up ok but there's no audio on 75 percent of the calls, so obviously a media issue (RTP) Now the complicated bit, how the network is setup The Fusion pbx has its own public IP on a...

Ping plotter against an endpoint over time will give you everything you need to know in 90% of situations. Run it with small samples, and UDP packet sizes that match RTP (or whatever your use case is). Run it for for as long as you can but for a minimum of 24 hrs. If you want to run VoIP on a...

This could be absolutely anything, you really need analyze both the network and your hardware as the problem is happening. You could just be running into network congestion on your LAN caused by something that happens at the same time once every 2 weeks or once a month. A backup, windows...

I've tested this on the Linksys SPA942 and the Cisco SPA504g