Well I tried maintenance mode - out of service for 10 mins, then back on - no go. I put the password back into the system - bam.. damn thing is registering again.

@scottalanmiller said in Getting Familiar with Sangoma S500 running with FreePBX as back end. . .: @JaredBusch said in Getting Familiar with Sangoma S500 running with FreePBX as back end. . .: @JasonMinard if you have not been on the FreePBX community today, you are simply out of luck with the Sangoma phones. https://community.freepbx.org/t/accessing-rest-contacts-from-horizontal-key-within-call/56374/10 Well that really sucks. Some of the REST-APP functionality is neat. But I find it slow to use.

While not related to the SIP function, I don't see any point currently in making a new thread. if you want to add/change things in ARS (Automatic Route Selection) look here You'll see places to add your new SIP trunk under > Facility Groups > P1500 (or your own numbering scheme) > Trunk Groups/Nodes in the right pane, right click and add.

@JaredBusch said in Metered SIP calling: @DustinB3403 said in Metered SIP calling: @JaredBusch said in Metered SIP calling: @DustinB3403 said in Metered SIP calling: @JaredBusch just taking this with a grain of salt, but the usage isn't including what they had to pay for the service. Correct? So their total would be closer to ~$500/month, right? WTF are you talking about? Per site the average spend prior to this was ~$500, correct? And now they are spending ~$200 so it's a $300/month savings. No? Doesn’t work that way. There were two bills totaling $1,000. Now there is a single bill averaging $200. Oh!! I completely misunderstood. So that's a much larger savings.

@eddiejennings said in Strange PBX CDR Entries: I've disallowed SIP guests. We'll see if I get future CDR entries like these. You won't.

@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.

@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.

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.

@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.

This really old post gives a clue on what the structure is for their firmware numbering. http://forum.yealink.com/forum/showthread.php?tid=1661 The template of Firmware name of Yealink phone is X1.X2.X3.X4 (1) X1 means the Phone Model X1- Phone Model T4 Series 35 - T48G 28 - T46G 29 - T42G 36 - T41P (2) X2 means the Major version number X2-Major version number 61 - V61 70 - V70 71 - V71 72 - V72 (3) X3 means the Custom or Neutral version 0 or 1 means Neutral version Non 0 or 1 means Custom version (4) X4 means the Minor version Yealink will release some minor versions after solving some bugs.

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.

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

This is the thread that pointed me to the Uniden equipment. http://forum.yealink.com/forum/showthread.php?mode=linear&tid=4681&pid=#pid

@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.

For another site I manage, I don't use stun and I also have ALG/SIP disabled on SonicWall but it is vlaned.

@StuartJordan Next time you decide to spin it up contact me and I will help you with the NAT issue. I have a little time this week. Next week I'm teaching FusionPBX November Admin training as show at www.fusionpbx.com. So if not this week I'll have more time after the training is over.

@biglittle said in The Basics of VoIP SAMIT Video: Just realized today when routing a ticket that our VOIP is pbx based? No idea what any of this means, but excited to learn Essentially all business VoIP uses a PBX. There can be exceptions, but it’s mostly theoretical.

@fateknollogee Agreed. Always good to layout what is available, let the people make a choice as to what is right for their situation.

@jaredbusch said in SIP Softswitch Real Addr vs Gave Addr: @gjacobse said in SIP Softswitch Real Addr vs Gave Addr: I am not sure how this will relate,.. but I am staring at our PBX and and this page. And then I see the different port numbers - It's interesting to see so many that are not 5060 No relation. this is NAT networking 101. All of your devices connect to pbx.ntg.co:5060 But their local outbound ports cannot all be 5060. How would the router know which devices to translate it back to. Yup,. I get that. - well anyway. moving right along...

Yeah it's not something we care for, as there are to many nuances for a system that has no tie in to what we need

@wirestyle22 said in Its to early to do math - SIP cost calculations - confirmation needed: I'm very tired and read this as "Is it too early to do Meth?" If you're awake, it's not too early

@travisdh1 said in ZOIPER unable to register over cellular: @dustinb3403 said in ZOIPER unable to register over cellular: It was apparently a setting under Connectivity > Run in Background With it disabled I'm able to register the account without WiFi. That is just so odd. Yeah, it was the only setting that made any sense under the connectivity option. Might be a bug, but seems odd that with it disabled, you can register the account. With it enabled and WiFi disabled, you can't register the account.

Same thing on the switches, prioritize that tag end to end.

Unlocked. Post about conference phones here.

@whoolly said in Switchvox phone issues: Vendor insisted he has never had any VOIP issues with Sonicwall and didn't want to budge on that. Even while it doesn't work. So you know that he'll say this to other customers now, even after this one. Chances are, he's had problems at all customers. SonicWall is culprit #1 for VoIP issues. I mean that literally. I get a call that someone has VoIP audio issues, my first question is always "Do you have a SonicWall?" Nine times out of ten, the answer is yes and nine times out of those ten, the SW was the issue. It's nearly a sure bet with audio issues. Had you led this question purely with "I have these audio issues..." we'd have said "I bet you have a SonicWall."

As far as many comments about redundancy and not all eggs in one basket. Just need to have a backup system that runs a nightly backup and restore script to another server or two. Setting up a server cluster is also possible for both systems.

@scottalanmiller is right. The bottleneck is always the customer router. And not so much bandwidth as much as packets per second. Your dropbox sync is killing your phone calls 50 to 1 over bandwidth 99% of the time. Your $50 linksys router can't handle a million pps. Most ISP's are little more than Broadsoft resellers with no interconnects in their local market. Even on the WISP side speeds are so good now that bundling voice isn't relevant.

@Mike-Ralston look at the PBX I have with NTG you can see outbound route there like that