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I haven't upgraded yet and this is a noob question, but I was editing : /usr/share/freeswitch/scripts/app/voicemail/resources/functions/record_message.lua. I made a change to my voicemail box in the gui and all my Watson edits got nuked. Should I be editing in a different location?

It's not really NAT its just what they call the keep-alive settings on that particular phone. I was thinking that if they're not getting a 200k OK back maybe the phone has a bad or closed off NAT state on the pfsense box. The keep alive is really more for UDP than TCP though.

Do you have nat mapping enabled per line? Seems simple but you may not be punching a hole through the Nat table

I should mention, I've worked around the problem with ring groups, ie follow me to ring group which then dials out. However at some point a user is going to want to set follow me in the gui.

The CPU spikes because of registrations is really weird. I'm currently migrating to Fusion but on our old asterisk systems we have several hundred SPA5XX phones registering at 300s intervals via UDP and we barely move the needle on CPU (relating to registrations). this is all done on bare metal...

I'm going into production soon and am going to need this to work or it could be a deal breaker for a lot of clients. I'm no coder but I'll be troubleshooting in earnest over the next week or so and will update the thread accordingly.

Great idea! It's always better to ban all and have a white list. I haven't run the script yet, but perhaps asking the user for a fail safe admin IP, that would be allowed no matter what would be a good idea to prevent lockouts. Just my 2 pennys

I actually am enjoying freeswitch more than asterisk. It's just some of the applications need to be tweeked.

This probably won't help but have you tried the simple stuff? Putting a pound sign at the end of the dial string in Fusionpbx won't do anything and I'm thinking that's why you're getting this: Maybe you have a pattern matching problem on your outbound route, can you actually dial 16102238642...

Cool, Just didn't know exactly what the procedure was to get the DBs into ram and Baulders way seemed to make sense, but if that's all it takes I'm in. Seems weird though that you can't change where the DBs reside.

Lol, I'm not production, but I put my own phone on it because there's not test like eating your own dog food. Everything looks OK, I was able to reproduce by going "uhhhh" in the test message. Unless Watson throws more strange characters we should be ok.

I'm running into this as well. I did everything Baulder did as well but the DB still ends up in the default path. I'm thinking about making /var/lib/freeswitch/db a soft link to my ramdsik, but my spidey sense is telling me that might be a very bad idea, needless to say kind of hackney. Any ideas?

It's hard to replicate as Watson is sending it as some sort of hesitation detection code, see example below: I've seen it happen in 2 messages though in the last 48 hrs so its a problem

Sometimes Watson will return an % which the send_email script will not properly escape causing this: 5c980fc2-3014-4ae0-9e6e-12807ec96998 2019-07-29 17:36:18.530976 [NOTICE] sofia.c:1079 Hangup sofia/LAN/6043432757@10.1.5.115 [CS_EXECUTE] [NORMAL_CLEARING] 5c980fc2-3014-4ae0-9e6e-12807ec96998...

Ok, so I RTFM and I did the following on my internal profile: inbound-late-negotiation = false disable-transcoding = true Basically if it comes in G711 it stays G711.

Hi, I have G722 setup as the primary codec on my system with G711 as the secondary. However I really want to avoid trans coding if possible. I'm going to try and set up all of my internal calling as G722, however I have some carriers that will accept G722 and I'd love to send and receive...

What hypervisor do you use?

Are you using rfc2833 in your sip profile and is the phone also using rfc2833? Also look in the log and check the byte length of the tones.

Are you accessing your voicemail by pressing *97 or *98 and then logging in?

Confirmed, I tried my original script from a G711 phone and it failed