@JaredBusch Thank you for that information

@fuznutz04 said in Site to Site VPN - not passing audio traffic properly: This one was interesting to get to the bottom of. @JaredBusch With the VPN tunnel enabled, the phone system was trying to send RTP to the phone on the internal IP. There is a setting in FreePBX on the extension level called "RTP Symmetric". Normally, this is set to yes. I changed it to no and the audio started flowing normally. However, I didn't like this solution. So, as a test, (and what I should have done from the beginning) I blocked all outbound traffic FROM my phone system, to any local network. (10.x, 172.16, 192.168, etc) This immediately solved the issue. I did not yet do a packet capture AFTER the fact to confirm, but I am assuming that blocking the PBX's ability to get to an internal private IP, forces the system to renegotiate and send the RTP to the correct public IP. Definitely an odd issue. nice you found a solution - I'm curious why it happens in the first place? Are some of the original phone's packet data still containing the original IP? And if so, why? Are you using encrypted RTP?

@JasGot said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail: @scottalanmiller said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail: We can't limit in Skyetel, or the calls won't make it to the voicemail. My first thought would be to route calls (when their are 4 current calls) to a specific DiD that goes right to VM. Are you working with Skyetel with custom programming or are you limited to their web portal options? Not sure what we'd have them do to fix the issue.

This is not an Asterisk error, so much as a FreePBX error. More information needed.

said script [[email protected] ~]$ cat unpause_all.sh #!/bin/sh set -e rasterisk -x "queue show ${1}" | grep paused | grep -o Local.*/n | while read -r member do rasterisk -x "queue unpause member ${member} queue ${1}" done

@JaredBusch said in FreePBX update negated/erased an Asterisk Dial Code we had set?: @Dashrender said in FreePBX update negated/erased an Asterisk Dial Code we had set?: yes, this would drive my users insane. It is a setting. Turn it off. I wouldn't want it off whole sale. I could definitely do that for most users, but my phone does have a display large enough to tell me I have missed calls - and I call those people back, even if they didn't leave a message. More often than not it's beneficial to call them back and solve whatever problem they were having.

@JaredBusch said in Incoming Call Issue: @VoIP_n00b said in Incoming Call Issue: Running FreePBX v14... No resent changes made, except to change the name on an extension. When some calls in, and let's say they dial 555 then end up getting 511, 511 can then forward the call to 555 no issue, but I have no idea what is causing this. Because they released a horrible fucking update to IVR. Edit your IVR and set strict dial time out to No - legacy That fixed it! Thank you so much!!!

@quintana said in Wazo to sponsor Astricon 2019: @JaredBusch I have a small demo for creating your own call control with our API. Find the source code here, it's in python, but i only use the rest api an websocket in the platform https://github.com/sboily/wazo-demo-programmable. I'm using the application API. http://www.wazo-platform.org/documentation/api/application.html#tag/applications Will look into it later this week. thanks.

Typo in the port, we found it.

Very nice. That’s one thing that’s seriously missing from FusionPBX!

@JaredBusch said in Extensions not registering: Fully updated FreePBX and zero issues. I use pretty much only use pjsip. This is VitalPBX, its super weird still. I didn't really touch anything on the extensions.

@hobbit666 said in Exploring VitalPBX: Do you know if they plan on changing Not much choice, chan_sip is deprecated.

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.