@pete-s said in Yealink T41P and T41S difference?:

I've actually found one difference and that is that the T41S has a USB port and I don't think T41P has one.

Additionally, the general difference between the G and S line was the USB port and the ability to handle the OPUS codec. I would assume the T41P did not have OPUS support either.

@rickygm now based on your experience, what other options are there to push with asterisk? I don't want to depend on third party services.

Thanks @JaredBusch

@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.

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@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.
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@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.

@Dashrender said in DVX 8000 IP PBX CONFIGURATION:

Awesome to see more telephoney stuff show up.

Bad telephony, but meh. all good.

@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.

At least the notification requirements are pretty simple and straightforward. For almost everyone, just email out a notification.

Your definition is not relevant. The FCC definition is what will be enforced.

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@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 🙂