@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]

@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.

@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.

3cx_compress_recordings.png

@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.. 🙂

@Dashrender said in New PBX - on prem or off?:

Damn I just saw a bill from cox.

A single sip phone totally managed by them end to end, $51.00/m

That includes:
The phone
DID
Vmail
Management of the line from the phone all the way to cox.
Calling inside the USA, ie no long distance fees.

Very expected when you buy from an ISP.

@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.

@eaglex said in FreePBX Contact Manager to Yealink Address Book:

Hello,

First of all, let me thank you for your wonderful work. This made our lives much easier...

If you don't mind, I would like some help: I'm running this script on 2 FreePBX machines, both are practically the same. On one machine, I have no problems viewing contacts with more than one number (In both the XML file and on the phone itself).
However, on the other machine, I can only see on phone number for each contact, even if they have more than one phone number, again, it happens both when I visit the XML file directly from a browser and on the phones.
I can't think of any difference between both the pbx's.

I'd appreciate your help and I'll be happy to provide additional details, logs, info, etc..

Thanks a lot!

My fingers are still crossed for a solution 😞

Thanks!

@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.

The problem with doing a fwconsole ma listonline on a new install is that it will immediately attempt to then install some commercial modules that have severe security updates.

And those also have dependencies that get updated.

I have not had a chance to come back around to find the best solution for an initial loop.

Also, more experiments proved out that you can simply do the delete.

@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.