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 [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.
Inbound: 5,649 at $0.0045 / minute ($25.42)
Outbound: 6,421 at $0.007 / minute ($44.95)
Inbound: 4,363.9 at $0.009 / minute ($39.28)
Outbound: 5,376.3 at $0.01 / minute ($53.76)
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.
Also have you tested your ping to to NJ/NY data center for VULTR? Might surprise you.
I was surprised and ended up moving some things from Chicago to NJ/NY despite it being twice the distance from me as Chicago VULTR instances were (500 miles to Chicago, 1200 miles to NJ/NY). Latency went from 80/90ms to 30ms or less on average when I moved to NJ/NY Vultr instance.
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.
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.
How do you secure this for truly mobile use (crazy guy is allowed to demand the use of a physical phone from anywhere)?
Ideally, crazy long passwords, and better with TLS on the SIP channel. You can go further with port knocking and similar. VPN is an option. But good passwords go a long way, and adding TLS goes really far.
You don't hotdesk with secure creds though because you cannot trust users type all of that in. That is why it is a bad idea for this scenario.
The best option is simply to use secure credentials like always and just allow the extension to authenticate from more than one device.
This requires training the users to push their ext button before dialing out when using the shared phone in the office.
This will likely be the way I'll handle the folks who come into the office once per week. This project is the perfect opportunity to introduce new behavior.
@Dashrender Right now, there is no security outside of extension number and passcode. Out-of-office users, simply have their phones configured to talk to our on-premises server and they login using their extension's credentials. Everything's in the clear.
Hence your toll fraud. I can log on to any of those extensions from anywhere.
@EddieJennings I don't know about the third, but if you start a hosting trial at freepbxhosting.com you get the first two free. Even if you then mode your install to Vultr it seems the licenses follow you for free.
@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.
ok I'm going to ask, but I know this will cause trouble. What about fax on FreePBX; or should we use cloud services? (he says crouched down behind his desk waiting for the onslaught)
FreePBX has a built in fax module. It can receive faxes no problem - it's sending that I'm not sure it can do.
I am personally curious how many incoming faxes it can handle before it has issues? My office gets around 650 pages of faxes a day, there was a concern around ML that FreePBX might not handle that well.
@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).
Original phone on original port and verify behavior.
Original phone on new port and verify behavior.
We had the original phone on the original port that was malfunctioning as described in the OP. We replaced the phone and got the same result. The new phone, which had the same exact behavior as the original (so it's not the phone) was moved to another port and had the same issue plus the lack of audio on the other end of a call.