@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 : 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 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 Glad it worked as intended.

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.

@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.

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.

@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.

You may also want to look into cable management solutions as well.

@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.

github updated. files renamed a bit. Still a work in progress, but things working much better.

0_1533163657304_1cb7dbcf-38f3-48da-a102-fc1f7348fe17-image.png

The downside to this is that the FreePBX team simply implemented the no reboot flag to the existing reboot-yealink command. unlike the multiple commands for the Sangoma phones.

0_1543282841223_bc79c9df-ec75-492b-a23b-d976e99dd85a-image.png

So this means, you have no way to reboot Yealink phones at all with a built in sip notify command.

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.

To Do: Find a way to make that the default setting on a new extension.

@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.

@fateknollogee said in Looking at Asterisk Operator Panel options:

Voice Operator Panel

Interesting. It seems to be a full softphone though. I will definitely add it to my list though.

0_1523543243826_451d9852-d8e0-443a-ae89-062c071a90e1-image.png

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

@jimmy_k said in Urgent: How to fix this FreePBX Repo Access Issue?:

@jimmy_k Well, I just fixed it
I modified etc/resolv.conf by putting this

nameserver same as DNS1
nameserver same as DNS2

That's what that script is supposed to do. If you did this, it means either you never activated the script or it is broken and will stop working again.

@bbigford said in FreePBX Reload Failed Retreive_Conf Encountered Error 1:

@scottalanmiller said in FreePBX Reload Failed Retreive_Conf Encountered Error 1:

Shut down Asterisk manually, restarted again with fwconsole start. Seems okay now. Odd that a reboot didn't pick that up.

That is weird... Was the service not running after the reboot?

Oh it was running. It just wasn't happy.