Good note, but horrible formatting.

@scottalanmiller I was just about to ask if its a php setting that needed to be changed.

so what we did was 3 step process - put the extension number on first alternative, another on second alternative, edit the call re-routing for the said extension number pointing to the line of 1st and 2nd alternative. we use the current timer but instead of going to VM it goes to the alternatives.

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 VOIP Provider: Skyetel: Well then, I stand corrected. I will add this is not something you have ever hinted at doing I. The past. Not mentioned a setup for (I would like to see that as I have no opportunities to use it yet). I only manage the phone bit of it, had nothing to do with the writing of the application. But it is neat what it does. The "users" (employees on our side) need to do a bit of fancy calling on behalf of their customers and the application handles setting everything up. It is quite involved. It has to be anonymous to one side, and has to handle non-voip scenarios on the other. So the application actually calls both parties most of the time. And handles automated recording and data collection and fills in a database with details and a copy of the call to go into a report to the customer.

@jaredbusch said in FusionPBX Install Error on CentOS 7: @fateknollogee said in FusionPBX Install Error on CentOS 7: Just emailed Mark at Fusion. He said "Yes, the official FusionPBX install script works fine with Debian 9 and Freeswitch 1.8." So outdated documentation? Or what? Seems like just one page of it wasn't updated. The rest was.

@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.

@scottalanmiller said in Public STUN servers: I tend to use: stun.stunprotocol.org This one.

@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.

@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.

@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.

@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.

@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...

@jaredbusch said in At What Size Should a Business Have a PBX: @travisdh1 said in At What Size Should a Business Have a PBX: @jaredbusch said in At What Size Should a Business Have a PBX: @travisdh1 said in At What Size Should a Business Have a PBX: 2-3 seems large when I'm setting one up just for myself, now that I know my cell phone randomly refuses to go to voice mail WTF? Yes, exactly my reaction when I heard about this. Waiting on my tech support request response. Just got that in today. No, I mean the first part. WTF does that even mean. Confused, late night postings apparently.

@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.

@dashrender said in Let's Convince Someone to release a FOSS PBX: @bigbear said in Let's Convince Someone to release a FOSS PBX: Assuming they moved to a support only model for revenue, would they have to charge an arm and a leg like XO to cover their bases, basically driving the support cost out of most SMB anyway? LOL @scottalanmiller said in Let's Convince Someone to release a FOSS PBX: @dashrender said in Let's Convince Someone to release a FOSS PBX: @bigbear said in Let's Convince Someone to release a FOSS PBX: When you consider the exposure these guys are missing vs the small amount of revenue they get from their baseline PBX products... how many times a week do you tell someone to use FreePBX? The question is how small is it? Do we really know? Assuming they moved to a support only model for revenue, would they have to charge an arm and a leg like XO to cover their bases, basically driving the support cost out of most SMB anyway? XO doesn't have to, their investors make them. Not the same thing. Not true. It was more a seed raising. edit: hahahahahahaha. I think I just understand, XO meant not "Xen Orchestra" Sorry for the confusion it didn't really made sense Sorry

@jaredbusch said in Issabel PBX " Elastix Alternative ": @scottalanmiller said in Issabel PBX " Elastix Alternative ": @alyragab said in Issabel PBX " Elastix Alternative ": I think that the purpose of the Issabel is to continue creating the Add On Modules that was supported by Elastix , also it is based on CentOS 7 which will be more reliable than what Elastix 2.5 was. anyway i think we need to wait till we find Issabel is stable distro and we can depend on it as we did with Elastix. I assume that they are working form the nearly released Elastix 4 work. That was on CentOS 7 at the end. And it did work with a lot of patience and knowledge of the underlying systems. But it was a hard fail for the layman. True. I had it working, but it wasn't ready for prime time when they gave up on it all.

@scottalanmiller said in DID Block Puchases: @Danp said in DID Block Puchases: @scottalanmiller said in DID Block Puchases: That's just weird. How so? Each DID is independent of the B channel, so I can have a multitude of DIDs and only a few B channels if so desired. Except it's a weird pairing. If you wanted flexibility to sell one at a time. Not "almost every channel." But it is a did not a number assigned to a channel.

@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.

I'm going to try this out. Looks interesting.

I believe that @QuixoticJeremy is doing a talk about something kind of similar.

@TeleFox we have a quite old (few years) PBX which has been used in the years adding new and new phone lines with the company expansion. last batch of additions was 6 months ago: 5 new seats with new phones and headsets. PBX is Aastra with proprietary digital phones. the company bought a new erp last december, against my advice they also buyed a generic "VoIP-ERP integration package". we are now implementing the ERP and the sysadmin at the ERP consulting firm pointed out to me that the package actually is a proprietary VoIP PBX based on asterisk. the solution involves something around 10 or 15 people (depending on the layout of the customer care dept), company has somethin like 40 phones bwteen DECT and digital deskphones. so we can: throw away the new asterisk PBX (wasted 4k€), sorry 15 people throw away current Aastra PBX , redoing a lot of cabling and rebuying the entire equipement for 40 between wirede and DECT phones try to integrate the 2, especially considering than it is right to isolate the ERP asterisk thing as you would not to depend on an erp for your phone system (ok, redo VoIP wiring and equipement if you want but choose a different strategy!) now both current PBX managing firm and the carrier have found a number potential solutions (basically a sip trunk between the 2 PBXes). hope this clarifies!