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.

You can use HTTPS, but you need to be aware that Yealink phones have issues with Let's Encrypt certificates unless you are on one of the most recent firmware versions of the current S series phones (T42S, T48S, etc.).

@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.

@JaredBusch said in FreePBX commercial module cleanup: @bnrstnr said in FreePBX commercial module cleanup: @JaredBusch Yes, I did. I've seen the error before, but not everytime like you say. I verify it is no installed as you did, and then delete it Ok, it looks like it's working, I was just curious about the error. No problems deleting them. Thanks

@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.

@romo said in Errors in outbound calls in one extension - Freepbx: This would be an inbound call right? You have a bad inbound route setup (you have ports forwarded and no catchall any/any route) or you allow SIP guests. This is a call terminating on your system but to with a DID you do not own. This is a probe.

@jaredbusch Ok thanks for the explanation

@jnaugle also worth noting, no one actually delivers PRI today, that entire market is a scam. It's actually SIP and then they translate the SIP to PRI on your network edge to make it "sound cool" for people who are out of date and remember when PRI was both still available and actually practical (early 1990s and earlier.) PRI is not nearly as powerful as SIP, nor as cheap, so phone providers long ago "all" moved to providing SIP instead (PRI isn't possible without a physical T1) and "virtualizing" PRI on top of it. All the PRI does it increase cost and lower performance, flexibility and reliability. So the best solution is to get the phone provider to stop pretending to do PRI and just give you the raw SIP. This improves your phone quality by removing unnecessary steps and limitations, removes the need for you to convert back to SIP (gateway, card, however you do it) and makes everything better for everyone. The PRI interface has no advantages, only disadvantages. It takes the better SIP, and cripples it. For people running ancient phone systems from before VoIP, SIP isn't possible, so these terrible PRI interfaces are useful for them to continue to be able to hook up those ancient phone systems while still being able to connect to current phone companies. They aren't totally useless, but they are useful only for that one terrible niche case.

@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.

@bigbear said in Manually Updating FreePBX: @jaredbusch said in Manually Updating FreePBX: @bigbear said in Manually Updating FreePBX: @jaredbusch said in Manually Updating FreePBX: @scottalanmiller said in Manually Updating FreePBX: @bigbear said in Manually Updating FreePBX: @scottalanmiller said in Manually Updating FreePBX: @bigbear said in Manually Updating FreePBX: So at this point I would assume a clean install of FreePBX 14 with Asterisk 14 is recommended? I've had better luck with Asterisk 15. So FreePBX14 with Asterisk 15? That's what we are using. Yes, FreePBX 14 with Asterisk 14 is what I have everywhere. I was thinking this. 14 and 14. Have you used FreePBX14 with asterisk15 in production? All of my production FreePBX 14 are Asterisk 15. So now that everything is fixed and the update is complete I see the update software section of System Admin is gone. I had read that they can automatically manage security and updates now. Is that what you are doing, or did you disable it? I prefer to do it myself because I don't want to deal with a down PBX. Granted that has never happened from a normal update, at least to me. It is now under Admin -> Updates It has this scheduler, but it does not seem to actually do anything except check.

@obsolesce I think that is what I had to do a couple years ago once. No idea why it started doing it and couldnt figure out any other way.

@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.

@jaredbusch said in FreePBX 14 super slow to update and then reboot: I did not post a bug report there yet as I have no idea if this is reproducible. I mean I reverted to the snapshot prior to the update and then did all of this. But I reverted because I had exactly this same problem after the update last night. But I would expect a system to perform the exact same under this scenario.

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.

/sigh. The expected answer. I know that EPM is a huge pain in the ass. But that is also why I willingly pay for it.

Made no configuration changes to the firewall tonight. Shutdown FreePBX VM, made a new one, and stuff seems to be working as it should. I'll do a few more tests tomorrow to make sure all is well. Edit: I lied. I made a new DHCP reservation for my new FreePBX server.

@dashrender said in FreePBX - Determine if Extension has dedicated callin: Look at your inbound routes. See what the destination on the inbound routes are. Yup, that extension itself doesn't know what points to it. So it is looking at the pointers. Think of the Inbound Routes like aliases and you are looking to see if an alias mentions that extension.