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The log you are showing is the part after it goes through the dialplan. "State Change CS_ROUTING -> CS_EXECUTE", is where it ends the dialplan and starts executing any matching actions. So we can't see if it is matching the correct dialplan from what you posted. However, based on what it does...

Set order to 90 and retest. Then if still not working post a new log snippet.

Is carrier receiving the request? (If not, then it isn't making it out of your server. If it is, then the carrier might be rejecting it.) Try using sngrep to see SIP activity. I'm not seeing that it's executing a bridge statement, which is normally at the end of the outbound route. For...

It looks like maybe you have your phones registering on the "external" SIP profile instead of the "internal" one? All phones should be registering on internal profile (usually port 5060). It looks like it's treating the call as inbound, which explains why it fails (you don't have a destination...

It could be a lot of things, but start with the obvious: Did you create outbound route (and the corresponding gateway)?

VERY good points!

For call recordings, we only have one customer where we keep it longer, so we adjusted the script like so: find /mnt/nfs/recordings/*/archive/* -not -path "*/<exception domain>/*" -name '*.wav' -mtime +$days_keep_call_recordings -exec rm {} \; find...

It works well for the providers that have been added to it, and for what it is intended. It isn't designed for sending/receiving from the web GUI, but it works well for receiving to endpoints (softphones) or email and sending from endpoints (softphones). Inbound MMS is implemented for some...

If you’re referring to the SMS app, it’s more geared to softphone usage, not to web-based usage.

That is the official "Messages" app. I have no experience with that. All of my development and usage has been on the SMS app. @markjcrane would be better able to advise on this.

Setting P2 works fine for us, no particular requirements on complexity that I know of. (we have over 50 of these units deployed, either ht812 or ht814, with a few ht818s as well.) I do not recall seeing the message about the backup firmware. I haven't been in the GUI of a brand new one in a...

Try manually putting the config on the device (download from Device page, then upload in the ATA's web GUI). This will confirm whether there is an issue with your template. If it uploads and registers successfully, then you know the problem is the file name, or some rewrite setting in Nginx...

Yeah, if you search that log (from "serialize") and do not see the agent name you expect, then it likely isn't being included in the event info. So then you have two options: 1. It MAY be possible to use a dialplan setting to insert the agent name as a custom channel variable that WOULD be...

When you run the "serialize" command to the log, do you see the agent name in any variable? If so, use that instead of "directory_full_name" or "variable_directory_full_name". If not, the CC_Agent_UUID is not the actual agent name. It's just a the lookup key to find that agent's info. In...

In my experience, sngrep will also show registration attempts. But maybe there are different versions. Sorry, I haven’t used WebRTC a lot, but have had to use various tools to troubleshoot registration problems from endpoints. Comparing the logging/PCAP from the endpoint to the output from...

You might also compare different types of endpoints to see if the issue is specific to webRTC/sip.js. Try a reputable softphone or desk phone or maybe a different webRTC client.

Well, maybe someone else can chime in here who is more of an expert, but it looks to me like there is some issue with sip.js properly performing the authentication every time, consistently. I would recommend that you run sngrep on your FreeSWITCH machine and compare the output there with what...

Not related specifically to WebRTC, but running sngrep on the FreeSWITCH side might provide some insight into what's happening when it connects.

I have not done much programming in this area, but I would examine a CDR record and see if it contains the name you are looking for, and then use the corresponding variable in your script. I went back and looked at one of ours and it had "directory_full_name" associated with the name. So this...

We started with sharing one domain for this type of customer, but as Adrian said, if they later add additional service, it’s a hassle. Also some things like call limiting and 911 routing and even user logins are more complicated by this design. We’re in the process of moving them to individual...