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That's correct. The User/Pass fields are used for gateways that register and require authorization. Also, since BW never sends a 401 or a 403 in reply to an INVITE the authentication fields are not used. You are authenticated by your IP address.

I have IP based SIP trunks to bandwidth.com. I built 6 gateways in fusionpbx. 2 for incoming, 2 for outgoing, and 2 for emergency routing. Bandwidth.com will provide the termination IP's during the setup. Incoming calls are E.164 so you need the "+1" when you build your destinations. Good luck.

Everything I'm seeing from Google makes it look like crypto incompatibility. Certs, client OS, OpenSSL, browsers, mod_verto all come into play. I did see someone say something about having FS & web server certs that match. Does a FS restart fix it? Wondering if Let's Encrypted updated the certs...

Does it work with this webrtc phone? https://www.doubango.org/sipml5/ Perhaps a certificate or openssl issue on the server. Any OS updates lately? Also, I see G722 as the codec. Can you try something else? Mine connects using OPUS right now.

Teams client -> teams server -> MS Graph Server -> Custom middleware that talks to MS & then injects events into Freeswitch -> FreeSWITCH -> BLF What could possibly go wrong :) I'm guessing that once you receive a status change from Teams you would need to call a lua script with parameters...

I'm pretty sure you cannot get a Teams User's status over SIP. You must get it over an API. They support notification subscriptions too.

We have known for a while that an easy way to screen robo calls is to answer the call with an IVR. Most automated calling systems don't know to press a digit to connect to the appropriate callee. Only a real person will know to do this. In this example i've replaced the call_screen dialplan step...

In order for hold to work properly, you need to strip out the "HOLD" method from the "Allow" header. If Microsoft thinks that you support hold it won't work. I put this in my route to handle replies from freeswitch. This was the trickiest part to get right. t_on_reply("handle_allow_methods")...

@mydigitalself, Good catch. I'll look at that on my end too.

File. Memcache is the legacy method and is not recommended.

Fail2ban just parses log files for specific keywords. If your log files are rapidly growing then fail2ban will run more or take longer to run. You may want to watch your logs to see which one is the offender. If it's the FreeSWITCH log, consider setting the log level lower. So if it's set to...

Try this... https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files resume-media-on-hold When calls are in no media this will bring them back to media when you press the hold button. To return the calls to bypass-media after the call is unheld, enable...

I've played around with VoiceOperatorPanel and liked it pretty well. It's a soft phone designed for operators. It's more of a client side rather than service side approach for operators.

So when you put " /var/lib/freeswitch/recordings/audio.wav" in the box it doesn't work? Perhaps check the file permissions. Should be owned by 'www-data'.

Interesting...if the file isn't on the file system then you aren't going to hear anything. You need to upload the file, find the file on the file system and enter the full path in the "Announce Sound" field.

See if your "find" command works on a known file. Also, Recordings uploaded from the web interface typically live somewhere in /var/lib/freeswitch/recordings/.

find / -name 'my_recording.wav'

I use streaming for MoH. I use Icecast as the media streaming server, and Ezstream as the source client. Each stream requires an instance of Ezstream to be running. It works well, but the setup of each stream is a manual process with coping mp3's, creating playlist m3u's, ezstream cfg's, and...

A couple of thoughts. Do a packet capture of the full call (SIP + RTP). Best if you can do this directly on the server or a mirror port of the server. In Wireshark, select Telephony -> RTP Streams, then Analyze. Look at the Graph. Do you see big jumps correlating with the dropped audio? Also...

Sessiontalk? There's an app for that.