Recent content by KonradSC

This position is filled. Thanks everyone!!

Hello, We are looking for a full time Sr level FusionPBX engineer. The person must be based out of the US and can work remotely. This is for a existing FusionPBX installation with approximately 100 customers and 25,000+ endpoints. Must have solid working knowledge of the parts and pieces that...

You could try adding it on the domain-variables dialplan step.

I would not modify that one. Keep in mind that the hook I sent is for extensions too. So adding another hook in that dialplan step is not going to work for things like ring groups.

There is an api_hangup_hook already in the dialplan. You can use this as a jumping off point. https://github.com/fusionpbx/fusionpbx/blob/master/app/dialplans/resources/switch/conf/dialplan/890_local_extension.xml#L18

Yes. In my config I have 2 public nics and 2 private nics. That config is for failover, not for an aggregate link. FreeSWITCH support was not very keen on a port channel and I had trouble getting that working. A 1 gig or 10 gig interface with a failover should be plenty anyway.

This is a very divisive topic. :) Android vs iOS, Chevy vs Ford, Meat Eater vs Vegetarian. Metal vs Virtual. We moved to bare metal for our primary servers years ago and haven't looked back. I was grew tired of questioning the hypervisor every time there was some kind of blip in call quality or...

I am running a similar setup and have not experienced this issue. I have some servers on Deb 8 and some on Deb 10. Right now we have about 350,000 recordings in the db. One thought...do you have any cron jobs that will scan your s3fs directories or scan all files. I'm thinking of the default...

From my understanding the requirements apply to interconnected service providers like Bandwidth.com, T-Mobile, or Verizon. If you are a hosted voice provider just managing phones and get your dialtone from someone like Bandwidth.com, they will take care of all the STIR/SHAKEN stuff and send...

Hello. Does anyone have any tips or tricks for getting SIP traffic to pass correctly through a Fortigate firewall? We are running SIP over UDP and have disabled SIP ALG & SIP session helper as stated in this KB article. https://kb.fortinet.com/kb/documentLink.do?externalID=FD36405 All phones...

Just hook a lua script to the end of the call using something like this: <action application="set" data="api_hangup_hook=lua app.lua email_recording"/> You'll probably need to pass the uuid of the call and the recording name & path to the script, lookup the email address, blah, blah, blah. I...

I use scripts that run fs_cli commands & sqlite queries. I have SNMP run these shell scripts and then chart the values using a monitoring application called PRTG. Example of my script to show the calls on the system: Example for registrations from sqlite Example for channels Example for...

Perhaps you could enable Follow-Me on extension 101 and enter the mobile number there. I haven't dug into the code for Click-to-Call in a while so I forget exactly how it's setting up the call, but follow-me could be a quick work around.

Can you grab a SIP trace from the call going out to the gateway? Also, do you see anything in the logs generated by the gateway?

Are you sure that extension 10000 is registered? This looks to be where you start to have issues.