@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!

@JaredBusch said in FreePBX - forward the main phone number when desired: from Peru (Country Code 51). lol nice.

@bnrstnr said in FreePBX fail2ban or SIP provider whitelist on router?: @scottalanmiller said in FreePBX fail2ban or SIP provider whitelist on router?: @bnrstnr SIP-ALG in ISP provided gear is nearly universal. Guh, who would have thought... Is SIP-ALG purely sabotage or is it useful in certain scenarios? Actually sabotage. Ubiquiti is the only vendor that I know where it works most of the time. Most firewall vendors are also either phone companies or in bed with phone companies and have a huge interest in convincing people that other phone products don't work reliably.

You can use HTTPS, but you need to be aware that Yealink phones have issues with Let's Encrypt certificates unless you are on one of the most recent firmware versions of the current S series phones (T42S, T48S, etc.).

@scottalanmiller said in FreePBX Error /var/spool/asterisk/cache is not writeable: These commands assume that you are logged in as root. If you are not, be sure to prepend with sudo. Or use root like a man, it'll put hair on your chest, even if you're a woman... so watch out I guess.

@scottalanmiller I was just about to ask if its a php setting that needed to be changed.

@scottalanmiller said in FreePBX commercial module cleanup: @JaredBusch that's a handy list, thanks. Well the instruction on how to do it was in the first post. As this can change overtime, that is important. Also , I choose to leave things like EPM and Zulu even if not currently being used.

@JaredBusch said in FreePBX Call Forwarding/Transferring: @Curtis said in FreePBX Call Forwarding/Transferring: Can you change the Asterisk Dial Options in a way that isn't system wide? Maybe just allowing it sometimes? That’s something that can be changed per trunk. So if you craft your rules to send those kind of calls out that trunk that would be the only time it applies. I was just thinking along those lines Thanks for your help

@JaredBusch said in How to add a sip notify command to FreePBX 14 to force Yealink phones to reboot: After everything reloads, you can open up a ssh session and tell a phone to restart like this. asterisk -rx "pjsip send notify restart-yealink endpoint 110" You can also do more than one phone at a time. Just space delimit them . asterisk -rx "pjsip send notify restart-yealink endpoint 103 108 110"

I agree that moving to a new system right away seems like the best course of action given that the phones are not actually down, but just cannot continue to be configured.

@bnrstnr that's your most likely culprit, having it be SIP aware.

@romo said in Errors in outbound calls in one extension - Freepbx: This would be an inbound call right? You have a bad inbound route setup (you have ports forwarded and no catchall any/any route) or you allow SIP guests. This is a call terminating on your system but to with a DID you do not own. This is a probe.

@jaredbusch Ok thanks for the explanation

@jnaugle also worth noting, no one actually delivers PRI today, that entire market is a scam. It's actually SIP and then they translate the SIP to PRI on your network edge to make it "sound cool" for people who are out of date and remember when PRI was both still available and actually practical (early 1990s and earlier.) PRI is not nearly as powerful as SIP, nor as cheap, so phone providers long ago "all" moved to providing SIP instead (PRI isn't possible without a physical T1) and "virtualizing" PRI on top of it. All the PRI does it increase cost and lower performance, flexibility and reliability. So the best solution is to get the phone provider to stop pretending to do PRI and just give you the raw SIP. This improves your phone quality by removing unnecessary steps and limitations, removes the need for you to convert back to SIP (gateway, card, however you do it) and makes everything better for everyone. The PRI interface has no advantages, only disadvantages. It takes the better SIP, and cripples it. For people running ancient phone systems from before VoIP, SIP isn't possible, so these terrible PRI interfaces are useful for them to continue to be able to hook up those ancient phone systems while still being able to connect to current phone companies. They aren't totally useless, but they are useful only for that one terrible niche case.

@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.