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I notice Mark just added a new parameter today contact-in-ping that is required for TEAMS integration. Have you set that? Maybe post up everything you have done so far and i'll help with the testing as well.

W where did you get that information from?

It’s an asterisk feature where calls will attempt to route back to the agent the caller last spoke with (if that agent is available)

Something I've been testing recently is this http://voisonics.com/sipsentry/ for an extra layer of protection. In general I block all international destinations except for a few major countries and only open others on request, I use voipmonitor and have been slowly adding notifications in there...

I thought exactly the same thing but the software itself looked quite good although quite complex but after a few weeks of testing i'm now really starting to understand it and i'm actually very impressed with the feature set it's becoming easier to use. Feature wise it's on par with most of the...

Im still learning how to use it as with any billing system they are bit complex at the start. The integration piece is really simple, it's basically a one click integration and it imports all of your domains and starts importing the CDR's. So far I'm impressed they also made a change for me...

I just purchased neon-soft and testing it now. I'll let you know how it goes.

Hey Guys, Has anybody successfully setup the "track-calls" parameter on the SIP profiles so that you can failover to a secondary server and keep the call alive. In the SIP profiles I set "track-calls" to "true" and made sure both freeswitch servers had the same switch name in switch.conf.xml...

To get some more information: How many SIP profiles are you using for registration? Are they general business users or call centre or home users? What are your system specs? Is CPU or RAM close to maxing out? Version of freeswitch? What is the SIP registration expiry timer? Are clients using...

Did adding the sound file fix the issue?

Flush cache, restart SIP, reload xml try all them as well

I would setup a call centre queue and have all the deacons as agents. Give them access to be able to log agents in and out. You don't want to be getting calls every time a deacon takes the day off and want the call to go somewhere else and by giving them the power to log agents in and out that...

Something similar happens to me as well so if sending a call directly to a ring group I always play a "greeting sound file" just 500ms of silence to answer the call. I'm sure there's a better way than this but that's what I've been doing.

The FusionPBX team responded and advised that is just how it works. If you want the transcription you must select "attachment"

Just submitted a bug report

Tested and have the same issue as you

Here's the post on the freeswitch mailing list of somebody that has it working. I've also seen a few others on the maining list that have it working. https://lists.freeswitch.org/pipermail/freeswitch-users/2020-October/134047.html

A As of Freeswitch 1.10.3 there is a new parameter: [mod_sofia] Add new gateway param 'contact-in-ping' to support Microsoft Teams Direct Routing I've read on the Freeswitch forums that you can use this to route traffic directly between Freeswitch and Microsoft.

I'll test this today and confirm if it's happening on my system as well.

Ive never experienced any NAT issues for the past 10 years. I’ve done the 3CX training and they talk a lot of NAT issues but honestly they rarely even exist if you have a half decent router. The issue that you are more likely running into is where clients have SIP ALG enabled that’s where the...