Setting up a Skyetel SIP trunk in FreePBX
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
so that you don't rely on DNS. In older versions of Asterisk (we still have users who use 1.4!), DNS would cause Asterisk to crash.
This is specifically a PJSIP trunk. It cannot exist prior to Asterisk 12, and wasn't normal until Asterisk 13.
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
Additionally, some people use less-than-awesome DNS solutions that either take a long time to propagate or are just providing inaccurate results.
If they are that bad, they will have other support issues anyway. You cannot fix stupid.
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
We typically recommend creating multiple trunks with our IPs
That creates its own disaster of teaching people how to route outbound calls.
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@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
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@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
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@JaredBusch yeah I was only quoting @Skyetel.
But thanks for the props?
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
And what is wrong with that? It's only 7 years out of date. . .
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@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
so that you don't rely on DNS. In older versions of Asterisk (we still have users who use 1.4!), DNS would cause Asterisk to crash.
This is specifically a PJSIP trunk. It cannot exist prior to Asterisk 12, and wasn't normal until Asterisk 13.
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
Additionally, some people use less-than-awesome DNS solutions that either take a long time to propagate or are just providing inaccurate results.
If they are that bad, they will have other support issues anyway. You cannot fix stupid.
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
We typically recommend creating multiple trunks with our IPs
That creates its own disaster of teaching people how to route outbound calls.
We prefer "inexperienced" to "stupid"
We also do have the guide for outbound routing:
https://skyetel.atlassian.net/wiki/spaces/SUG/pages/16613436/FreePBX+Outbound+RoutesBut I do agree - this is a more elegant implementation than what we document. Thanks for writing it
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
Still use is one thing. Phone systems are still very legacy mindset about lasting forever with no maintenance.
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@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
Still use is one thing. Phone systems are still very legacy mindset about lasting forever with no maintenance.
Still use = actively deploy. We obviously discourage this lol
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
Still use is one thing. Phone systems are still very legacy mindset about lasting forever with no maintenance.
Still use = actively deploy.
Yeah, ok.. i would refuse to service them. Does that even run on CentOS 5? How woudl they even find it? All fo the old domains redirect.
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@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
Still use is one thing. Phone systems are still very legacy mindset about lasting forever with no maintenance.
Still use = actively deploy.
Yeah, ok.. i would refuse to service them. Does that even run on CentOS 5? How woudl they even find it? All fo the old domains redirect.
I assume they've kept the ISO for years. TrixBox isn't the only legacy system people still deploy - we have ancient Avaya, Mitel, and even Toshiba systems on our network. People tend to stick with what they know and are afraid of updating their PBXs. We do our best to help, but when they are using systems that old, usually they already know that they are going to have a hard time before they even contact our support department.
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@DustinB3403 said in Setting up a Skyetel SIP trunk in FreePBX:
It uses SRV records instead of A records.
One would assume that any modern OS can handle that.
We seriously do have users who still use TrixBox.
Still use is one thing. Phone systems are still very legacy mindset about lasting forever with no maintenance.
Still use = actively deploy.
Yeah, ok.. i would refuse to service them. Does that even run on CentOS 5? How woudl they even find it? All fo the old domains redirect.
I assume they've kept the ISO for years. TrixBox isn't the only legacy system people still deploy - we have ancient Avaya, Mitel, and even Toshiba systems on our network. People tend to stick with what they know and are afraid of updating their PBXs. We do our best to help, but when they are using systems that old, usually they already know that they are going to have a hard time before they even contact our support department.
Heck, we still have an old ass inter-tel.
Hope to change that either this summer or next.
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@Dashrender said in Setting up a Skyetel SIP trunk in FreePBX:
Hope to change that either this summer or next.
Not holding my breath
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@JaredBusch said in Setting up a Skyetel SIP trunk in FreePBX:
@Dashrender said in Setting up a Skyetel SIP trunk in FreePBX:
Hope to change that either this summer or next.
Not holding my breath
yeah - sadly, neither am I - I'm guessing they might go the cheaper route and simply replace the digital brain with the modern/supported unit, and keep the rest of the infrastructure intact...
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@Skyetel said in Setting up a Skyetel SIP trunk in FreePBX:
We typically recommend creating multiple trunks
VitalPBX's guide also uses a single
pjsiptrunk in the same method I did here.They posted a guide a couple days before I did this one, apparently with the assistance of Skyetel staff.
So you might want to rethink your "recommendation"
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One thing to clarify here (I will also put this info in the first post).
By default, @Skyetel does not use the e164 standard for the inbound DID format, they use an 11 digit number. There is nothing wrong with this, but it makes it more work to "standardize" the displayed number to your users. I'm a huge fan of never showing the country code. So for US numbers that means stripping the 1.
FreePBX does not have anything built into their inbound dial plan code to strip a 1 fomr an 11 digit number. But they do have a process to strip the +1 from a number of any length. That is the
from-pstn-e164-uscontext I say to use in the guide.Fortunately, @Skyetel makes it very simple to send the DID in the e164 format.
Select your phone number(s) and click edit.
Right on the
Generaltab you will see SIP Format.
Click on the drop down and change it to
+1NXXNXXXXXXand click save.
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The benefit (to me) of setting this up like this means that your inbound routing is all just 10 digit numbers.
