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

I can't think of a way aside from cron every X seconds. I'd dump it into a csv file with just the timestamp and the number of calls. E.g. #!/bin/sh if [ ! -f /var/log/activecalls.csv ]; then echo "Timestamp,Calls" > /var/log/activecalls.csv fi DateTime=`date "+%Y%m%d %H:%M:%S"` echo -ne "\n$DateTime," >> /var/log/activecalls.csv asterisk -x 'core show channels' | grep 'active calls' | cut -d " " -f1 Which SHOULD create a nice csv for you...

@bnrstnr said in Asterisk Usage Survey: If this survey was a test, I think I failed Wow - yep, a ton of functions I don't know or use specifically.

@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.

@dragon3303 said in Using a Snom PA1 for paging: Yes...I've tried adjusting the level but no results with that either. I'll keep at it and figure it out. Thanks for pointing me in the right direction. I would have to Google a manual for your amp to see what the specs are for these inputs and such. I was strictly assuming based on the labels.

@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.

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. 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.

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.

@jaredbusch Thank you

@jaredbusch said in SIP Desk Phones Not Re-Registering with Main WAN's IP After WAN Fail-back: @scottalanmiller said in SIP Desk Phones Not Re-Registering with Main WAN's IP After WAN Fail-back: @taurex said in SIP Desk Phones Not Re-Registering with Main WAN's IP After WAN Fail-back: @jaredbusch Thanks for this link. I'll give it a try. I hope UBNT did not trim the CLI down too much on the USG. I now better understand why Edge series is still the prefered choice for deployment among many Especially now that UNMS is available, that's pretty slick. UNMS 0.12.0 (currenlty in alpha) will add the CLI to the web interface. That's very exciting. So much good stuff happening with UNMS.

@stuartjordan said in Yealink T19PE2 FreePBX: I personally like freepbx and will continue using it, the interface is a bit dated I must admit, but that doesn't bother me. FreePBX interface is definitely dated. But also .... why would the interface ever matter that much? But even if it did, I've used 3CX and their interface was far worse.

@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.

@jaredbusch said in FreePBX CDR and CEL Analysis: @eddiejennings A unique ID is a call. How wold you think that it is not a valid representation of calls? Yeah. . . I was having a dumb moment yesterday. I've actually taken a step back and asked the user to review with me what information they're actually wanting. I'm taking the angle of "now that we have FreePBX, let's see if [your information needs] can be better met."

I pulled a @jaredbusch and justed used voip.ms

I'm going to try this out. Looks interesting.