I just found this. I guess it is called "Camp" or "Call Back" https://en.wikipedia.org/wiki/Automatic_callback Not I can start researching how to do it with VitalPBX. The conversation here, on this topic was instrumental in getting me on the right path for a solution. Thank you all! We can keep talking about it too, if you want

Typo in the port, we found it.

We still publish with yellowbook. They give us stats on how many people search for us online.

@scottalanmiller said in Best SIP VoIP Softphone for macOS: Got a couple of users on macOS that would like to set up a softphone on their desktops. In theory Linphone, Zoiper, X-Lite make macOS versions. Anyone have a reason to pick one or the other or have another option on macOS? I use Telephone on my laptop and like it. Its pretty sparse, but it works well and is efficient.

@JaredBusch said in Prerequisites for Yealink YDMP Install: @DustinB3403 worked with this I believe... I did, but the install process was pretty easy. https://drive.google.com/open?id=1-2hh-5N-vg-s2w8MS3GVaqDRVetLmbpp

@hobbit666 said in Exploring VitalPBX: Do you know if they plan on changing Not much choice, chan_sip is deprecated.

@black3dynamite said in FreePBX Desktop Video Conferencing Options: Have you already check with linphone if it does video conferencing? In theory it does. Linphone desktop has been pretty flaky for us so were looking at alternatives.

@Dashrender said in Relationship Pricing: @JaredBusch said in Relationship Pricing: But with the second one there, I only perform a lookup once every 30 days for numbers that call me. Is this because FreePBX caches the info? Yes, but any system can be setup that way.

@JaredBusch said in Make Yealink Ring While on the Phone: @scottalanmiller said in Make Yealink Ring While on the Phone: If you have Yealink T46S phone, and you are talking on the phone, instead of making it do a call waiting beep, can you make it ring normally? A bit odd, but there is a reason for it Not to my knowledge. It was supposedly added as a feature request a few years ago, and I have not noticed it in a firmware release note. That would explain why I have never seen it come up. Not that anyone has ever wanted it previously.

@wrx7m happy to help!

Yeah, outdoor speakerphones arent a popular item.

@JaredBusch said in Call Out from FreePBX Conference Room: @scottalanmiller said in Call Out from FreePBX Conference Room: @JaredBusch said in Call Out from FreePBX Conference Room: Hit transfer, hit the conference extension, dial the participant code, speak their name, hit the transfer button again. The idiot will be in the conference. I've tried that, always seem to get stuck with the "enter a code" issue. You are going to force me to test those instructions aren't you? Tested what I typed above.. Works perfectly.

Well I tried maintenance mode - out of service for 10 mins, then back on - no go. I put the password back into the system - bam.. damn thing is registering again.

@JaredBusch said in Getting Familiar with Sangoma S500 running with FreePBX as back end. . .: I've given up on EPM for anything complicated with non Sangoma phones. Same, I don't find it to be valuable for us.

While not related to the SIP function, I don't see any point currently in making a new thread. if you want to add/change things in ARS (Automatic Route Selection) look here You'll see places to add your new SIP trunk under > Facility Groups > P1500 (or your own numbering scheme) > Trunk Groups/Nodes in the right pane, right click and add.

@JaredBusch said in Metered SIP calling: @DustinB3403 said in Metered SIP calling: @JaredBusch said in Metered SIP calling: @DustinB3403 said in Metered SIP calling: @JaredBusch just taking this with a grain of salt, but the usage isn't including what they had to pay for the service. Correct? So their total would be closer to ~$500/month, right? WTF are you talking about? Per site the average spend prior to this was ~$500, correct? And now they are spending ~$200 so it's a $300/month savings. No? Doesn’t work that way. There were two bills totaling $1,000. Now there is a single bill averaging $200. Oh!! I completely misunderstood. So that's a much larger savings.

@eddiejennings said in Strange PBX CDR Entries: I've disallowed SIP guests. We'll see if I get future CDR entries like these. You won't.

@jaredbusch said in Asterisk 13 CRITICAL ERROR! Required Service CHAN_SIP is disabled! This device is unusable!: does chan_sip.so exist in /usr/lib/asterisk/modules/? My guess is, no. But once we determined that there was no simple restoration and all evidence of chap_sip was gone from the system, they just rolled back to a snapshot from before the outage and all is well. So we might never know.

@scottalanmiller said in All Yealink Phones Down at One Site: @dbeato said in All Yealink Phones Down at One Site: @scottalanmiller said in All Yealink Phones Down at One Site: @i3 said in All Yealink Phones Down at One Site: Factory reset a phone and try manually programming? How about a packet capture at the site firewall, do you see attempts from that IP phone to where? I am assuming that the PBX is hosted elsewhere. This is basically where we went. We did this series of events and it appears to be fixed... Rebooted the firewall. Removed SIP-ALG, but this appears to have had no effect. Updated the firmware on each phone. Went to static DNS rather than DHCP set. DNS to CloudFlare instead of AD. Ensures STUN was on and set correctly. Added the PBX as a Proxy, manually enabled STUN for the proxy. After those seven steps, all phones are now registered. Couldn't find any step there that individually made a difference. To me looks like the DNS was caching something for the devices to get to. That's my theory. If so, why did changing the DNS alone not fix it? Not sure, I am just going by what you said. Basically very strange.

We still have the error, but we did determine that it's just an error message and the clients can register fine. The registration issue was unrelated. So not a big deal, but it would be nice to know why we get this.

@tm1000 said in FreePBX 14 Module Update Errors: Our QA team has confirmed that firewall has no issues: https://community.freepbx.org/t/bug-in-latest-firewall-adding-ip-to-firewall-trusted-zone-no-longer-updates-iptables-firewall-rules-on-firewall-restart/50181/9?u=tm1000 I've got a few on the latest and haven't seen anything ourselves, yet. Seems good.

This really old post gives a clue on what the structure is for their firmware numbering. http://forum.yealink.com/forum/showthread.php?tid=1661 The template of Firmware name of Yealink phone is X1.X2.X3.X4 (1) X1 means the Phone Model X1- Phone Model T4 Series 35 - T48G 28 - T46G 29 - T42G 36 - T41P (2) X2 means the Major version number X2-Major version number 61 - V61 70 - V70 71 - V71 72 - V72 (3) X3 means the Custom or Neutral version 0 or 1 means Neutral version Non 0 or 1 means Custom version (4) X4 means the Minor version Yealink will release some minor versions after solving some bugs.

Made no configuration changes to the firewall tonight. Shutdown FreePBX VM, made a new one, and stuff seems to be working as it should. I'll do a few more tests tomorrow to make sure all is well. Edit: I lied. I made a new DHCP reservation for my new FreePBX server.

It's my first time running into it. Wondering if it might not be a corrupted client.

This is the thread that pointed me to the Uniden equipment. http://forum.yealink.com/forum/showthread.php?mode=linear&tid=4681&pid=#pid

@bnrstnr said in Weird Network Errors on SIP Trunks, FreePBX: @scottalanmiller We still use the legacy wholesale thing. It probably wasn't as often as every 3 months, I'm sure I'm exaggerating a little. Oh okay, that's what we are using, too.

For another site I manage, I don't use stun and I also have ALG/SIP disabled on SonicWall but it is vlaned.

@StuartJordan Next time you decide to spin it up contact me and I will help you with the NAT issue. I have a little time this week. Next week I'm teaching FusionPBX November Admin training as show at www.fusionpbx.com. So if not this week I'll have more time after the training is over.