Correct - you should be using PJsip. Also as far as changing extensions, you can run into the same issue even if you use the same extension number but you change the channel driver from SIP to PJSIP. This is how I've accomplished the task for remote phones that you can't touch. From your EPM, issue a reboot. Then real quick, you change the extension to use PJSIP, go to EPM, and then do a rebuild. If you do it properly, the phone will then grab the new settings and reconfigure itself and work just fine. You can of course do the same thing if you are changing the extension number on it.

Good way to make me not care about answering you.

@JaredBusch Thank you for that information 🙂

@fuznutz04 said in Site to Site VPN - not passing audio traffic properly:

This one was interesting to get to the bottom of. @JaredBusch With the VPN tunnel enabled, the phone system was trying to send RTP to the phone on the internal IP. There is a setting in FreePBX on the extension level called "RTP Symmetric". Normally, this is set to yes. I changed it to no and the audio started flowing normally. However, I didn't like this solution. So, as a test, (and what I should have done from the beginning) I blocked all outbound traffic FROM my phone system, to any local network. (10.x, 172.16, 192.168, etc) This immediately solved the issue. I did not yet do a packet capture AFTER the fact to confirm, but I am assuming that blocking the PBX's ability to get to an internal private IP, forces the system to renegotiate and send the RTP to the correct public IP.

Definitely an odd issue.

nice you found a solution - I'm curious why it happens in the first place? Are some of the original phone's packet data still containing the original IP? And if so, why?
Are you using encrypted RTP?

@JasGot said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail:

@scottalanmiller said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail:

We can't limit in Skyetel, or the calls won't make it to the voicemail.

My first thought would be to route calls (when their are 4 current calls) to a specific DiD that goes right to VM.

Are you working with Skyetel with custom programming or are you limited to their web portal options?

Not sure what we'd have them do to fix the issue.

This is not an Asterisk error, so much as a FreePBX error.
More information needed.

said script

[jbusch@pbx ~]$ cat unpause_all.sh #!/bin/sh set -e rasterisk -x "queue show ${1}" | grep paused | grep -o Local.*/n | while read -r member do rasterisk -x "queue unpause member ${member} queue ${1}" done

@JaredBusch said in FreePBX update negated/erased an Asterisk Dial Code we had set?:

@Dashrender said in FreePBX update negated/erased an Asterisk Dial Code we had set?:

yes, this would drive my users insane.

It is a setting. Turn it off.

I wouldn't want it off whole sale. I could definitely do that for most users, but my phone does have a display large enough to tell me I have missed calls - and I call those people back, even if they didn't leave a message. More often than not it's beneficial to call them back and solve whatever problem they were having.

@JaredBusch said in Incoming Call Issue:

@VoIP_n00b said in Incoming Call Issue:

Running FreePBX v14... No resent changes made, except to change the name on an extension.

When some calls in, and let's say they dial 555 then end up getting 511, 511 can then forward the call to 555 no issue, but I have no idea what is causing this.

Because they released a horrible fucking update to IVR.

Edit your IVR and set strict dial time out to No - legacy

That fixed it! Thank you so much!!!

@quintana said in Wazo to sponsor Astricon 2019:

@JaredBusch I have a small demo for creating your own call control with our API. Find the source code here, it's in python, but i only use the rest api an websocket in the platform https://github.com/sboily/wazo-demo-programmable. I'm using the application API. http://www.wazo-platform.org/documentation/api/application.html#tag/applications

Will look into it later this week. thanks.

Typo in the port, we found it.

Very nice.

That’s one thing that’s seriously missing from FusionPBX!

@JaredBusch said in Extensions not registering:

Fully updated FreePBX and zero issues. I use pretty much only use pjsip.
b6a3844c-2573-4763-bd7f-efce2b7603ae-image.png

This is VitalPBX, its super weird still. I didn't really touch anything on the extensions.

@PitzKey said in Exploring VitalPBX:

Phew! Wow, I just spent quite some time reading through almost every comment in this thread.
Fast forward to now, it seems like most issues were addressed.

Worth mentioning that we have been using VitalPBX in single and multi-tenant mode since I think early 2019 and have struggled with a ton of issues, but we are glad to see the progress they have made.

This was a good throwback ride!

Yeah, we use it heavily too.

Okay, we got it. Doing the switch, then the fwconsole update, then rebooting, and running through it all again... eventually it fixed itself.

I can't think of a way aside from cron every X seconds. I'd dump it into a csv file with just the timestamp and the number of calls. E.g.

#!/bin/sh if [ ! -f /var/log/activecalls.csv ]; then echo "Timestamp,Calls" > /var/log/activecalls.csv fi DateTime=`date "+%Y%m%d %H:%M:%S"` echo -ne "\n$DateTime," >> /var/log/activecalls.csv asterisk -x 'core show channels' | grep 'active calls' | cut -d " " -f1

Which SHOULD create a nice csv for you...

@bnrstnr said in Asterisk Usage Survey:

If this survey was a test, I think I failed :frowning_face:

Wow - yep, a ton of functions I don't know or use specifically.