@mmicha Yup, it's a good solution for sure. Nothing is perfect, but as long as you have CommQuotes by your side, you will have someone there to help you if you need it. We have clients with a few phones to hundreds of seats, as well as call centers that push millions of minutes per month over their network. You are in good hands and I think you are going to save so much money that you will be hoisted up on to a chariot and whisked away to the kingdom that awaits you as King of Phones at your company

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

@scottalanmiller said in Remote FreePBX Extension: @fuznutz04 said in Remote FreePBX Extension: @gjacobse said in Remote FreePBX Extension: NTG uses a responsive firewall on its internal PBX and have very very little rare issues. Set the phones server, stun, and such and you are good to go. You use STUN with the responsive firewall ? I haven’t had to use STUN for a while now, but I know a lot of people still have to use it. Just curious as to why? We use STUN always. The responsive firewall doesn't affect the need for STUN in any way. STUN is for the end points, not the server. Does the traffic from the endpoint via STUN not also appear form the STUN IP to the server? I rarely use it. so just wondering.

@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.

@smitherick Thanks for your suggestion I just tried with Reid method and it is working well. The red notification has been removed

@jaredbusch Thank you

@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.

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

@jaredbusch said in EdgeRouter "Smart" QoS issue: @fuznutz04 said in EdgeRouter "Smart" QoS issue: @jaredbusch said in EdgeRouter "Smart" QoS issue: @fuznutz04 said in EdgeRouter "Smart" QoS issue: The ER-Lite boxes have a "Smart" QoS wizard. The way I've always understood it, it works as follows: You fill in the line speeds of the service you are actually paying for. Example: The above screenshot is from my home, so the speed in the boxes doesn't line up with the rest of the details here, it's just and example. So if you pay for 100/10, that's what you put in the boxes. I do this at the office, at home, and other clients. However, I ran into a situation where using the Smart QoS wizard, actually really limits the DL speed. I did a speed test with the QoS setting ON, and got the proper UL speed, but was limited to 50-60 DL speed. As soon as I removed QoS, I got the proper speed of 100 DL and 10 UL. Has anyone experience this before on the Edge Routers? Perhaps @JaredBusch ? This is normal and expected because traffic policies and the smart queue policies both force traffic to not be offloaded. The CPU in the ERL cannot process traffic faster than ~60mbps without the offload functionality. Well alright then. Simple explanation. I’ve never run into this, because we have such slow speeds around here. So this would be an example where I could upgrade to the next model. Perhaps the ER4? I haven’t read the spec sheets yet on them. Correct using a different model is what is required to obtain better speeds in this situation. The ERX will perform better because it has a better CPU than the ERL. But it has less memory which leads to other problems. The ER4 is light years ahead of the ERL by comparison due to the newer CPU. OK great. Looks like I'll need to start recommending these now. First, I want to get one for myself at my office. Did you manage to get your hands on one yet?

That's an idea.

@dustinb3403 said in VoIP Bandwidth: @coliver said in VoIP Bandwidth: @dustinb3403 said in VoIP Bandwidth: @coliver said in VoIP Bandwidth: Welp... 7% packet loss looks like I need to call the phone company. Phone company...? (Everyone else sees it, right?) DSL... Ha.. "what are packets" is what I expect them to ask Pretty much. I've got a few contacts in the backend I hope I can tap to bypass the support hell that they contracted out to.

@scottalanmiller said in FreePBX 14 Firewall Start Warning: @jaredbusch said in FreePBX 14 Firewall Start Warning: @scottalanmiller said in FreePBX 14 Firewall Start Warning: Well, we are one step newer, so that might be it. # fwconsole ma list | grep firewall | firewall | 13.0.46.1 | Enabled | AGPLv3+ | I do not know the CLI command to revert, but it is simple to do in the GUI. CLick "Check upgrades and then expand the Firewall and you will have a previous versions option. Tested and you are right, rolling back to 45.5 and the message goes away. You could upgrade to edge and see if it is different, but I would just wait for the next update.

@dbeato said in Are VLANs Needed for VoIP? SAMIT Video: @scottalanmiller said in Are VLANs Needed for VoIP? SAMIT Video: @dbeato said in Are VLANs Needed for VoIP? SAMIT Video: @scottalanmiller said in Are VLANs Needed for VoIP? SAMIT Video: @dave247 said in Are VLANs Needed for VoIP? SAMIT Video: @scottalanmiller Didn't you already post this a while back? I searched and didn't find it. I don't think it got a thread, but it was likely a response to someone somewhere. Yeah, you posted it in SpiceWorks. I think maybe here, too, though. Maybe this? https://mangolassi.it/topic/15518/the-basics-of-voip-samit-video/34 Maybe that is all that I am thinking of.

@stuartjordan said in FusionPBX: implementing/using ??: How much time have you spent on FusionPBX Jared? Not enough. @stuartjordan said in FusionPBX: implementing/using ??: and what is pushing you to make the move? As I stated, it is a good solution, but I do not earn money from anyone to support FusionPBX. So it has always been a low priority solutoin to get more educated about. But I am now confident enough with the current FreePBX offerings that I do not feel the need to run our internal system on it in order to maintain skills.

@coliver said in Recommended FreePBX Commercial Modules?: Not necessarily an add-on from shmooze but FOP2 is phenomenal. This is a great tool, really cheap and you can use it on Freeswitch as well. Also easy to change the GUI to a more modern interface.

@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.

@coliver said in Fax Service: @dashrender said in Fax Service: @coliver said in Fax Service: @dashrender said in Fax Service: @eddiejennings said in Fax Service: I'm testing voip.ms's email to fax service. One immediate challenge is training users to send it plaintext E-mails or get them to delete all of the images from our default E-mail signature. >( Is it possible to have the email server strip that when sending to a specific address? Depends on the email server. Office365/Exchange Online can be configured to forced to send plain text emails to specific addresses IIRC. What happens in those cases when you have a PDF or a Word doc attached? Attachments would still be attached I believe. Could be wrong never had to do it. I can speak only to RingCentral's service, but if you have text in the body, whether it be a signature or logo or anything in the body, it will try to print that extra text out on a separate page in the fax. Looks messy. We initiate ours through email and we have a fax cover letter template that users can fill out electronically and then also attach whatever they're wanting to send, preferably in PDF format. So generally it's sent to ### @ rcfax.com, and then 2 attachments, one for cover letter and one for the actual item being sent. Putting something in the subject line messes things up as well.

I remember sitting with a Broadcloud rep 18 months ago and they were worried about how SFB on Office 365 would affect their business. Still find it amusing that we took it as a serious threat. I still have a lot of hope for FusionPBX and some other open source products as well.

@dustinb3403 said in VoIP over Wifi: You should set traffic shaping on wireless the same as you would on your LAN. Doesn't work the same. It's a bus, not a switch.

@jaredbusch said in Choosing a SIP Provider - What Should I Look For?: @dashrender said in Choosing a SIP Provider - What Should I Look For?: @wrx7m said in Choosing a SIP Provider - What Should I Look For?: I am checking out voip.ms and they have a number portability tester. I tested several and they seem to be OK. What would make the numbers not portable? asshole vendor No, that is not how it works. The carrier holding the existing number has no legal recourse to deny a port. Actually they do. We have carriers that do just that. Not all numbers are regulated in the US, sadly.

@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...

Interesting. I would contact them. If the per minute plan is no longer available, this is not such a good service to recommend.

@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.

@eddiejennings said in Independant Contractors on a FreePBX system: @dashrender Logically, I shouldn't make that assumption. The assumption is based on a gut feeling fueled by some experience. I gotcha - if I'm a contractor, and that's not a spelled out benefit, and it's clear that the benefit is being removed because I'm no longer using my own phone, I'd be rather unhappy at the pay cut.

@dustinb3403 said in Confirming QoS for VOIP on Procurve 2848: Ok cool. Now looking at another switch 46 and 48 have a priority of 7, and 26 a priority of 4. (without an override). Should I set everything to be like below? 46-7 48-6 26-4 Doesn't really matter. If you have no traffic on any other DSCP tags those priorities never apply. Also does not matter much if you are not saturating the switch port in the first place.

@eddiejennings said in [SIP Pricing: How much are 11.338 milliseconds worth?] Twilio: New Jersey at 12.853 ms average ping Voip.ms: Atlanta (Atlanta2) at 1.515 ms average ping Still trying to figure out where you are getting Twilio New Jersey? It seems like you are talking about the VULTR data centers to your office, and not the VULTR data centers to the pops. Twilio is 35% cheaper on inbound and at least 15% cheaper on outbound. Your DID numbers and toll free costs are irrelevant. You need to dump the DID's you arent using, maybe keep 5 extra. If you can dump half thats $600 per year. On actual costs see below. You are clearly saving money now and any increase in call volume will only add to those savings. You could save on toll free with a couple other people, but for ease of administration I would keep it all at one place. And remember, those PRI DID's cost the CLEC anything. I had a guy trying to port 800 DID's to me once and we ended up issuing 150 of our DID's, forwarding theres for 6 months and dumping them. My DIDs cost me nothing, porting those 800 would have cost me $200 per month and he would have been paying $800 per month to us. Twilio Inbound: 5,649 at $0.0045 / minute ($25.42) Outbound: 6,421 at $0.007 / minute ($44.95) $70.73 VOIP.MS Inbound: 4,363.9 at $0.009 / minute ($39.28) Outbound: 5,376.3 at $0.01 / minute ($53.76) $93.08 And imagine these numbers in multiples. $700 versus $930. Everything I hear about Voip.ms is good, but Twilio's infrastructure is many times larger. They are already cheaper. I cant really speak to Voip.ms but have been meaning to give them a try.

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