Okay, if you want to talk storage, it's over here: https://mangolassi.it/topic/19657/revisiting-zfs-and-freenas-in-2019

@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX: @Dashrender said in Setting up a Skyetel SIP trunk in FreePBX: Hope to change that either this summer or next. Not holding my breath yeah - sadly, neither am I - I'm guessing they might go the cheaper route and simply replace the digital brain with the modern/supported unit, and keep the rest of the infrastructure intact...

@JaredBusch said in FreePBX / Yealink t42s / call park configuration best practice ...: The best use of parking is simply blind transfer via DSS key directly to a specific slot. No one wants to waste time and a button with a park key that you then need to wait for an announcement to know where the call went. That makes much more sense. I think I put the blinders on and focused too much on replicating the old system without thinking if there was a better way.

@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.

@BraswellJay said in FreePBX assign commercial module to installation?: @JaredBusch I think I figured this out. The thing I needed to do was to stop and then start FreePBX. I did : fwconsole stop fwconsole start When it started back I saw the following as part of the output : Updating License Information for xxxxxxxx where the x's were my deployment ID I'm thinking this must be where it went out and queried and saw I had purchased those two modules. Both SysAdmin Pro and the Endpoint Manager screens now look like your samples. Thanks for the help. One question. When you have purchased these through the GUI instead of the portal as I did do you normally have to restart or does it just figure that out and is immediately available? It is all handled in a reboot. I buy these during initial turn up time and as part of that, I will always perform a final reboot before begining the actual configuration. Because one of the turn up steps requires setting the timezone, and that is not applied fully until a reboot. Honestly, I simply assumed that you had already turned it on and off again. You could have used fwconsole restart or clicked the update activation in the web GUI also.

@Curtis said in Bria Mobile iOS and FreePBX: I tried a few hours with Bira this morning... Just a couple minutes ago I downloaded Acrobits Groundwire by Acrobits, and it instantly worked... https://itunes.apple.com/us/app/acrobits-groundwire/id378503081 That listing clearly states it uses push. So how does that work?

@scottalanmiller said in Setting up a Skyetel Trunk on FreePBX: Don't forget to set your Codec to G722. But always offer G711 We do our best to deliver calls over G722 and follow your preferred priority (based on what you put in your SIP), but when we can't get the call delivered over G722, we fall back to G711.

@BraswellJay said in FreePBX / Random loss of audio ...: When I get a chance this afternoon I may configure one and see if that at least resolves the issue, though I didn't think I needed STUN for this particular use case. So I haven't totally found root cause but it appears to be an issue with my windows profile on the machine I was running Linphone on. I installed Linphone app on my phone and everything worked perfectly. Installed desktop app on a spare laptop and everything worked perfectly. Logged in as different user on original machine and everything worked perfectly. Logged back in as myself on original machine and no go. I only need this for some testing. Once we get set up we'll be using Yealink phones so I may not look anymore since I can use my phone for the testing. Thanks to all for the help.

@JaredBusch said in Repo issues with Freepbx: @scottalanmiller said in Repo issues with Freepbx: Been going on for several days. No it hasn't. It has, we started getting the error several days ago.

@JaredBusch said in Yealink T46S Double Tap while on Conference call: @CCWTech said in Yealink T46S Double Tap while on Conference call: @JaredBusch said in Yealink T46S Double Tap while on Conference call: DTMF Is that on the ext in advanced? It's set to RFC4733 Both the Phone and Asterisk have settings for this. They need to match. They do, we checked that last night.

Awesome information about Backup & Restore and Filestore.

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

@WLS-ITGuy said in fwconsole creating many emails through our Barracuda mail filter: Just ran updates in FreePBX and now the queue buildup is back. Guess I'll have to fix it the right way Enjoy!

If we were going to give containers a number, it would be more like a Type -1 than a 3. Containers are lighter than Type 1, not heavier than Type 2. Still totally different. But Type 0, 1, 2 go from lighter to heavier.

@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.

@fuznutz04 said in FreePBX and DNS settings: @scottalanmiller said in FreePBX and DNS settings: @fuznutz04 said in FreePBX and DNS settings: @Scott says not to set it. You say to set it to 127.0.0.1 and the Vultr DNS. No, Scott says he doesn't use it, not that you shouldn't. Not the same thing But doesn't this just write it to the /etc/resolve file anyway? I could set DNS statically via the file, but I'm still unclear what the best practice is for FreePBX in regards to DNS. To not fuck with it unless you have a reason to. FreePBX is an appliance. A software appliance, but an appliance. You don’t fuck with it under the hood unless you have a good reason to do so.

I figure I'd just throw this out there just for the heck of it... For what it's worth, I use a Cisco ISR 2851 as an FXS and FXO gateway in my home lab and am very happy with it. You can nab them pretty cheap on eBay (either a 2851 or 2821 would work). You'd just need either two VIC2-4FXO cards or for the more adventurous an EVM-HD-8FXS/DID with two EM-HDA-6FXO daughter cards and something to terminate the RJ21 connector with (like a 66 block or a breakout cable)...the former is likely easier though

The timeline. 2019/02/27 09:23 - I posted on the FreePBX community. 2019/02/27 15:13 - Another user commented they also had this issue. 2019/02/27 16:31 - I submitted a support ticket. 2019/02/27 17:03 - Sangoma replied stating this was not any kind of known issue and I needed to install their SSH key package to let them check the server. 2019/02/28 08:52 I replied that I had added the SSH package and updated the firewall to let them connect. 2019/02/28 09:49 Sangoma replied that the files were missing on the remote server and showed the log error. 2019/02/28 10:05 Sangoma replied that an internal trouble ticket was created. 2019/02/28 15:58 Sangoma replied that the issue was resolved. 2019/02/28 19:26 I confirmed that the firmware was downloadable This is not any kind of response time that I can complain about. I do wish there had been known information on how to find the EPM download errors. I would have likely been resolved much quicker. That's why I posted here. Search engines can get this out there.

@NashBrydges said in Mass upload sound files into FreePBX: Very useful stuff, thanks for this. Self serving. I’m buildinging a mass import right now using this process.

For reference, here is the same IVR dialplan as output by FreePBX asterisk -x "dialplan show ivr-4" [ Context 'ivr-4' created by 'pbx_config' ] '*' => 1. Set(__ivrreturn=0) [extensions_additional.conf:4167] [ivrsel-*] 2. Goto(app-blackhole,hangup,1) [extensions_additional.conf:4168] '0' => 1. Set(__ivrreturn=0) [extensions_additional.conf:4164] [ivrsel-0] 2. Goto(ivr-4,s,1) [extensions_additional.conf:4165] '9' => 1. Set(__ivrreturn=0) [extensions_additional.conf:4170] [ivrsel-9] 2. Goto(um-vm,s,1) [extensions_additional.conf:4171] 'fax' => 1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [extensions_additional.conf:4149] 'h' => 1. Hangup() [extensions_additional.conf:4192] 'hang' => 1. Playback(vm-goodbye) [extensions_additional.conf:4194] 2. Hangup() [extensions_additional.conf:4195] 'i' => 1. Set(INVALID_LOOPCOUNT=$[${INVALID_LOOPCOUNT}+1]) [extensions_additional.conf:4173] 2. GotoIf($[${INVALID_LOOPCOUNT} > 3]?final) [extensions_additional.conf:4174] 3. Set(IVR_MSG=no-valid-responce-pls-try-again&custom/client_main) [extensions_additional.conf:4175] 4. Goto(s,start) [extensions_additional.conf:4176] [final] 5. Playback(no-valid-responce-transfering) [extensions_additional.conf:4177] 6. Goto(app-blackhole,hangup,1) [extensions_additional.conf:4178] 'return' => 1. Set(_IVR_CONTEXT=${CONTEXT}) [extensions_additional.conf:4187] 2. Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT_${CONTEXT}}) [extensions_additional.conf:4188] 3. Set(IVR_MSG=custom/client_main) [extensions_additional.conf:4189] 4. Goto(s,start) [extensions_additional.conf:4190] 's' => 1. Set(TIMEOUT_LOOPCOUNT=0) [extensions_additional.conf:4151] 2. Set(INVALID_LOOPCOUNT=0) [extensions_additional.conf:4152] 3. Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT}) [extensions_additional.conf:4153] 4. Set(_IVR_CONTEXT=${CONTEXT}) [extensions_additional.conf:4154] 5. Set(__IVR_RETVM=) [extensions_additional.conf:4155] 6. GotoIf($["${CHANNEL(state)}" = "Up"]?skip) [extensions_additional.conf:4156] 7. Answer() [extensions_additional.conf:4157] 8. Wait(1) [extensions_additional.conf:4158] [skip] 9. Set(IVR_MSG=custom/client_main) [extensions_additional.conf:4159] [start] 10. Set(TIMEOUT(digit)=3) [extensions_additional.conf:4160] 11. ExecIf($["${IVR_MSG}" != ""]?Background(${IVR_MSG})) [extensions_additional.conf:4161] 12. WaitExten(10,) [extensions_additional.conf:4162] 't' => 1. Set(TIMEOUT_LOOPCOUNT=$[${TIMEOUT_LOOPCOUNT}+1]) [extensions_additional.conf:4180] 2. GotoIf($[${TIMEOUT_LOOPCOUNT} > 3]?final) [extensions_additional.conf:4181] 3. Set(IVR_MSG=no-valid-responce-pls-try-again&custom/client_main) [extensions_additional.conf:4182] 4. Goto(s,start) [extensions_additional.conf:4183] [final] 5. Playback(no-valid-responce-transfering) [extensions_additional.conf:4184] 6. Goto(app-blackhole,hangup,1) [extensions_additional.conf:4185] Include => 'ivr-4-custom' [pbx_config] Include => 'from-did-direct-ivr' [pbx_config]

@scottalanmiller said in Getting Familiar with Sangoma S500 running with FreePBX as back end. . .: @JaredBusch said in Getting Familiar with Sangoma S500 running with FreePBX as back end. . .: @JasonMinard if you have not been on the FreePBX community today, you are simply out of luck with the Sangoma phones. https://community.freepbx.org/t/accessing-rest-contacts-from-horizontal-key-within-call/56374/10 Well that really sucks. Some of the REST-APP functionality is neat. But I find it slow to use.

@JaredBusch Good thing you brought this up. I couldn't believe that your recording where ADPCM because in our 3CX we just have PCM recordings. It turns out that the ADPCM codec for recordings is a completely new feature in 3CX, which arrived in the latest version 15.5 I had to dig around before I could find that setting anywhere. But it's under the recording quota. IMHO you should have a lot of recordings to enable it in the first place. In just 10GB you have room for 167 hours of non-compressed recordings. But maybe some users have it set up to recording everything on every call and a lot of extensions and small cloud hosted VMs.

@JaredBusch : Funny that you posted that as I actually had to go LOOK UP NANPA when you had mentioned it above and, at first, wondered what it had to do with the Japanese dating scene..

@Pete-S said in New PBX - on prem or off?: @JaredBusch said in New PBX - on prem or off?: @Pete-S said in New PBX - on prem or off?: Makes sense. I'm mostly familiar with 3CX and it actually seems to be the opposite there. The options for each extension are: PBX Delivers Audio (default off) Support Re-Invites (default on) Support 'Replaces' header (default on) Support reinvite on the endpoint does not mean the PBX has it enabled. But I do not know enough about 3CX to tell you without being in a system. I'm no expert so I don't know. The point of my original post was that the PBX requires very little resources but perhaps that is true only in the few cases where the audio doesn't flow through the PBX. It's like a soft switch (actually, it IS a soft switch) and it takes almost zero effort to pass the RTP packets around.

@Dashrender said in Skyetel Inbound SIP Trunk on FreePBX: @JaredBusch said in Skyetel Inbound SIP Trunk on FreePBX: @scottalanmiller said in Skyetel Inbound SIP Trunk on FreePBX: @JaredBusch said in Skyetel Inbound SIP Trunk on FreePBX: @scottalanmiller said in Skyetel Inbound SIP Trunk on FreePBX: Locking a port to a single IP is a trivial task for any gear, even consumer. I wouldn't go that far. I don't know much consumer gear that doesn't make that easy. Super easy tomport forward? Yes. Able to restrict to a specific source IP? No. eh - if you're only open for the trunk provider, then you can likely lock to one or a few IPs. if you need mobile support - that's another story. How? Where? Been a while since I was in a consumer router, but port forwarding screens had nothing about source IP.

@JaredBusch said in Alternatives to OpenVPN for FreePBX on cell phone...: @scottalanmiller said in Alternatives to OpenVPN for FreePBX on cell phone...: @JaredBusch said in Alternatives to OpenVPN for FreePBX on cell phone...: There has never been a promise or timeline made for the mobile apps. Anyone expecting anything on that front is operating in their own little fantasy. That's the main use case of softphones, though. Like 95% I would guess. Softphones for the desktop are way more niche. I totally disagree. Softphone on the desktop is by far the largest user base thanks to call centers. Softphone on mobile is a far second to that. Maybe overall, but for FreePBX? FreePBX is pretty rare in call centers, AFAIK.

The FreePBX wiki article had a few optional settings that can be used with either version of the Trunk PEER Details I showed above. qualifyfreqok=25000 transfer=no trunk=yes The default qualifyfreqok is 60000. This just makes it qualify more often. This is usually less important on a trunk than an endpoint that uses IAX2. If you are setting up an IAX2 trunk, you already have tested your internet connection and know it is stable. If not, call me.. I'm more than happy to consult... The transfer=no is the same as the SIP allow reinvite that hands off a SIP call letting the endpoints not pass the call through the PBX any longer. Not applicable to an interoffice trunk. The trunk=yes will shave a few Kbps off of each call after the first going over the trunk as it reduces IP header information. You can read all the gory details here. Basically it is not enough to worry about.

@JaredBusch THANKS!