@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.

@jaredbusch said in How to add ability to FreePBX to reload Yealink config without a reboot: ModEdit: corrected spelling. Hahahahahahahahahahahha

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.

Been working with Yealink support on this and after confirming this happening with the T46G and the T42G phones, their support upped the severity and I got an answer back to add the below setting to the configuration files. transfer.hang_up_after_success_trans = 3 I did so on the T46G, disabled the custom firewall rules, and had the operator repeat the transfer. No ghost ringback. I'll leave the custom firewall disabled and see if it stays good all day. I'll test in more detail over the weekend or on Tuesday, whichever time I have a tech on site.

@JaredBusch said in Wazo 16.16 First Look: Not seeing anything in the Wazo documentation on how to set this up either. In the early release, definitely is looking like it is a bit hard to use. It's early though, hopefully that stuff will follow soon.

@scottalanmiller fwconsole starts asterisk, and it makes sure it's started and running with the correct UID Or it should. It did last time I checked Edit: Saying that, I believe I was the last one that messed with that code, so if it's NOT working, it's probably my fault. https://github.com/FreePBX/framework/blob/release/13.0/amp_conf/htdocs/admin/libraries/Console/Start.class.php#L147

@tm1000 said in FreePBX 13 No Conference Room App: This was happening because for some reason the conbridge.conf file (or an include therein) was not being created correctly and thus because conf files were missing confbridge was never loaded. I believe this was because conferences was not originally included in the tarball for 13 but we fixed this so it should be OK???? Not sure... It was fixed this week? It was just six days ago that we saw it, but it's not been re-run since then.

@tim_g said in Jared Busch: VoIP: @scottalanmiller said in Jared Busch: VoIP: @bigbear said in Jared Busch: VoIP: @scottalanmiller said in Jared Busch: VoIP: @bigbear said in Jared Busch: VoIP: @scottalanmiller said in Jared Busch: VoIP: @stuartjordan said in Jared Busch: VoIP: @jaredbusch This sounds cool, was there quite a few videos to be uploaded? In theory, quite a lot. Looking forward still to my first mangocon... You’ve got a little bit to wait for that. Summer 2018? Spring 2019, Dallas Perhaps around a national holiday so my daughter has off school and I can make it a family trip to Dallas? May is the guess.

@scottalanmiller So since I posted the OP, I found out what the issue was. (or at least eliminated it by changing a setting.) On old versions of FreePBX, in Advanced SIP settings, in order to enable TCP connections, you had to manually type: tcpenable =yes In new versions, there is now a button to enable TCP. Having both settings enabled seems to cause these issues. Most likely, the config file for sip.conf gets messed up having these similar/identical settings. Removing the old entry eliminates the errors.

@quintana said in XiVO - Open Minded Telecom System: @scottalanmiller Can you give me more detail. How did you install it? It's on the other thread as well, but it is just a generic ISO based install on Vultr.

@thwr said in MC16: VoIP with Jared Busch Session: @JaredBusch said in MC16: VoIP with Jared Busch Session: @scottalanmiller said in MC16: VoIP with Jared Busch Session: @JaredBusch said in MC16: VoIP with Jared Busch Session: @thwr said in MC16: VoIP with Jared Busch Session: @scottalanmiller said in MC16: VoIP with Jared Busch Session: Myths of VoIP: No 911 Is not secure Requires dedicated Internet Needs special network configuration have to buy all new hardware (or any new hardware!) Seriously? ... Serious that those are myths of VOIP? yup. I hear those all the time. Same here. @thwr Just happened last night in fact when i gave a presentation to a SpiceCorp. Really sad. But there are so many myths (and half-knowledge) in and around IT... Seems to be the majority more often than not