@Skyetel said in Phone Outages for February 19, 2021:
Skyetel’s President is addressing this personally, and all options are on the table. He’s, uh, pretty upset.
Hahhhaahahahahhah
Sangoma has relased an updated (and likely final) statement.
https://www.sangoma.com/press-releases/sangoma-technologies-provides-update-on-ransomware-attack-expects-no-material-impact-on-sales/
The second paragraph has the relevant information from an IT point of view.
@AlyRagab said in Asterisk/Elastix setup in AWS:
Actually it is a POC level of environment. I am convinced already that Asterisk "As a legacy app" is so difficult to make it cloud native kind of application.
For me i am away from the VoIP field since more than 4 years now, so is there any other complete solution that can be scaled and hosted in AWS ?.
You are getting things backwards. You should be responding with "oh, AWS and cloud are the wrong approaches for VoIP". It feels like you are driving with a solution and trying to find a goal. Cart before the horse. A POC of a fundamentally bad idea is still a bad idea and a waste of resources.
Is the idea of scaling VoIP interesting? Sure. Can you produce a use case for it? Maybe. But it'll be a stretch.
Does anyone make a complete solution? I doubt it. There's no market for it. The only places that could make use of it are enormous players who would make their own solution (e.g. Twilio) in order to do something unique. If you are the kind of shop that needs this, you can't really use someone else's product. If you need to do someone else's product, we can safely assume it's the wrong product for you.
@VoIP_n00b said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@Dashrender said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@travisdh1 said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@scottalanmiller said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@travisdh1 said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@syko24 said in Changing Installation Location for MicroSIP Installed via Chocolatey:
@scottalanmiller said in Changing Installation Location for MicroSIP Installed via Chocolatey:
By default, installing MicroSIP using Chocolatey causes the install to go to an inappropriate data folder location and is only available for the admin installing user, rather than for the end users of the Windows system. Does anyone have any experience or ideas in changing the installation location so that system users can actually use MicroSIP when installed and maintained using choco?
Asking for @Dashrender and @romo as well.
I think this might work:
choco install microsip -ia "'/D=C:\SomeDirectory'"
choco install microsip -ia "'/D=C:\temp'"
Did install it to C:\temp for me
Is it working for non-admin users?
Yep, I can run the .exe from my C:\temp. You'll probably have to create a shortcut for a user tho.
yep, I just tried it - I pushed to c:\program files\microsip
and non admins can run it.
and you're right, the shortcut is put on the installing admin's desktop, and nothing in the start menu.
MicroSIP - open source portable SIP softphone based on PJSIP stack for Windows OS.
https://www.microsip.org/
It says right on the website it’s a portable app, why would you expect a desktop icon or anything on the start menu?
There is a portable version, sure. But there is also a non-portable version.
@JaredBusch
Everything works fine now.
The issue was a DNS resolution for devices within the LAN. I added an A record on the router for split dns. That solved the problem.
Thank you so much for your support
@JasGot said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail:
@scottalanmiller said in VitalPBX / Asterisk Limit Calls Then Go To Voicemail:
We can't limit in Skyetel, or the calls won't make it to the voicemail.
My first thought would be to route calls (when their are 4 current calls) to a specific DiD that goes right to VM.
Are you working with Skyetel with custom programming or are you limited to their web portal options?
Not sure what we'd have them do to fix the issue.
@pirho99 said in ISP & VoIP woes:
If someone in the neighborhood is using MoCA, you could put this adapter on the business' incoming cable line, before the first split or amplifier. It would block their interference. link
This won't make any difference though - because the signal has to leave the building, i.e. go where the line noise is, to get out to the rest of the world... so the problem would still be there.
@DustinB3403 said in Kari's Law and Special SMB Onus on IT Staff:
@scottalanmiller You just said that Enterprise customers can be forced to update their information, but smaller orgs get a pass.
I must be misreading what you've typed.
Yes, the CUSTOMERS get a pass, IT that supports small customers has to provide the information that they don't have control over. Enterprise customers have to provide the information from the company.
SMB customers get to not tell anyone when changes are made and blame someone with no power or way to get that information for certain.
@DustinB3403 said in Kari's Law and Hotdesking:
@scottalanmiller I would say, you should disable it if someone can't pick up any phone and call 911.
It IS disabled in that situation. So removes it from law coverage.
@akiliss said in Auto provisioning server for yealink T21P:
Hi @JaredBusch, Thank you very much for your reply, I tested a same configuration on Friday and it worked fine with TFTP.
Now Im trying to configure this with HTTP or HTTPS.
Thanks again
You can use tftp if you want. It works just fine. My issue with it is the lack of security.
@JaredBusch said in Incoming Call Issue:
@VoIP_n00b said in Incoming Call Issue:
Running FreePBX v14... No resent changes made, except to change the name on an extension.
When some calls in, and let's say they dial 555 then end up getting 511, 511 can then forward the call to 555 no issue, but I have no idea what is causing this.
Because they released a horrible fucking update to IVR.
Edit your IVR and set strict dial time out to No - legacy
That fixed it! Thank you so much!!!
We had the same problem with VitalPBX, and @ScottAlanMiller's suggestion, we made the Opus codec the highest priority and it resolved the issue for us. Maybe it is worth a try for you?
I just found this. I guess it is called "Camp" or "Call Back"
https://en.wikipedia.org/wiki/Automatic_callback
Not I can start researching how to do it with VitalPBX.
The conversation here, on this topic was instrumental in getting me on the right path for a solution. Thank you all!
We can keep talking about it too, if you want
@scottalanmiller said in Best SIP VoIP Softphone for macOS:
Got a couple of users on macOS that would like to set up a softphone on their desktops. In theory Linphone, Zoiper, X-Lite make macOS versions. Anyone have a reason to pick one or the other or have another option on macOS?
I use Telephone on my laptop and like it. Its pretty sparse, but it works well and is efficient.
We've been using Skyetel as our primary trunks for some time now, and most of our customers are either using them now or porting over to them in the near term. Service has been great, and very easy to use.
@JaredBusch said in Prerequisites for Yealink YDMP Install:
@DustinB3403 worked with this I believe...
I did, but the install process was pretty easy.
https://drive.google.com/open?id=1-2hh-5N-vg-s2w8MS3GVaqDRVetLmbpp
@Pete-S said in Discussing with management switching to VOIP ...:
@Skyetel said in Discussing with management switching to VOIP ...:
@Pete-S said in Discussing with management switching to VOIP ...:
@BraswellJay said in Discussing with management switching to VOIP ...:
No internet - no phones : this is an issue in that we don't have redundancy on our network link, but there has not been any network outage since the fiber was installed at this particular site several years ago. Not sure what I can do on this short of say we could keep a POTS line or two for emergencies but that of course starts to eat into the cost savings some also.
Maybe you can setup a 4G link as redundancy instead. Voip traffic doesn't require that much bandwidth so even if you only get 2 Megabit/s or something it is still plenty.
I would suggest against this - a lot of the 4G carriers do crazy things with NAT that SIP Trunks really really do not like. We run into CGN problems all the time:
https://en.wikipedia.org/wiki/Carrier-grade_NAT. It also can create issues with UDP Fragmentation.
But if you are using a hosted PBX is that really an issue?
You can mitigate it somewhat, but yea, its still an issue. The best way to get around it with a hosted PBX is to use a VPN between the phone and the PBX. That will bypass most of the NAT issues and keep the phones routing the way they should. You can still get dropped calls though.
@JaredBusch said in Building VitalPBX on Vultr Automatically:
@scottalanmiller said in Building VitalPBX on Vultr Automatically:
and be fully updated.
I noticed that the defualt CentOS 7 image was pretty much fully up to date before I ran the install script. Only thing that came down when I ran yum update -y was wget.
Yeah, it is more of a "just in case" something is out of date. I almost never see it not fully up to date when I fire it up.
@black3dynamite said in FreePBX Desktop Video Conferencing Options:
Have you already check with linphone if it does video conferencing?
In theory it does. Linphone desktop has been pretty flaky for us so were looking at alternatives.
@Curtis said in Bria Mobile iOS and FreePBX:
I tried a few hours with Bira this morning...
Just a couple minutes ago I downloaded Acrobits Groundwire by Acrobits, and it instantly worked...
https://itunes.apple.com/us/app/acrobits-groundwire/id378503081
That listing clearly states it uses push. So how does that work?
@Dashrender said in Relationship Pricing:
@JaredBusch said in Relationship Pricing:
But with the second one there, I only perform a lookup once every 30 days for numbers that call me.
Is this because FreePBX caches the info?
Yes, but any system can be setup that way.
