What you are describing is very typical. 

 

There are a lot of other details in their network that you haven't described
as well - type of switch, what do they do with the PRI on the other end,
etc.  But, like the rest, it's not important to what you are needing to deal
with.

 

ITSP's deliver PRI to an end customer via an IAD that terminates their SIP,
and provides a PRI to the customer along with Internet service on an
Ethernet port all day long.  I can assure you it works for tons of their
customers, and they probably have tons of experience to back up what they
are doing. Their issue is typically that they only deliver sip to PBX
products that they have "certified" as compliant with their switch.  You
shouldn't care about how their deliver it to you, as that is their issue to
deal with.  They give you a PRI, you need to convert that PRI to work with
your sipXecs product.  You have put a Mediant in the middle to do that for
you.  Configure it to integrate between the PRI and sipXecs and quit
worrying about the rest of it.   When you have your piece set up correctly,
and it doesn't work when you give it to them, then isolate the issue and
throw it over the fence to them.

 

Today, you have a PRI to worry about, and a Mediant 1000 configured to
connect to your sipXecs.

 

From: [email protected]
[mailto:[email protected]] On Behalf Of
[email protected]
Sent: Sunday, April 24, 2011 7:36 PM
To: sipx-users
Subject: Re: [sipx-users] Faxing doc?

 

On Sun, 24 Apr 2011 19:08:29 -0700, Todd Hodgen wrote:
> I think Mike is misunderstanding and mixing up protocols here.

I've always admitted I just get by on this stuff, I'm no voice/telco
engineer, it's just one of the many things I need to have a basic
understanding of. Indeed, I sometimes don't know the proper terminology.

> He states
> his calls are coming in over a PRI.  They are digital trunks, utilizing
> qsig or some other standard ISDN protocol, ran over PCM and terminates to
> your Mediant.

Everything is coming in over a full data T1.
The T1 runs into an Adtran 908C Total Access unit.
That unit outputs a PCM connection to the mediant.
I never see SIP from their network. I only see SIP from my mediant to my
network.
It's silly but they are sending Voice/Fax using SIP, then convert to PCM
into the mediant which I then convert back into SIP for use on the LAN. 
They don't provide end to end SIP in which case I would not have needed the
mediant, which is what I thought when that connection was going in, til the
last minute and by then it was too late.

Let's see if I can draw something up.

Internet Connection
 |
Provider---Metaswitch for Voice/Fax services using SIP
 |
 T1 Circuit
 |
Customer
 |
Adtran Router---PRI-Mediant
 |
Lan Switch


>They need to be
> converted to t.38 for sipxecs, which I assume your gateway is capable of
> doing (Licensing?).

I don't know if this device has that licensing, if it's needed. Does have
plenty of T.38 options though.

> Set Mediant to support T.38, and sipXecs will recognize the call if set to
> the correct alias for the extension you want it sent to.  It will convert
> T.30 to T.38.

I could give it a try and see what happens.

> Here is a good article to read -  http://en.wikipedia.org/wiki/T.38_ITU-
> T_recommendation

Came across that a while ago while looking around for more info, thanks.

> Mike, I would encourage you to read this article so you can speak with
your
> service provider at a higher level and understand what is being setup on
> their end.

I've always spent time reading up on the things I'm working on. I wish I
could have better knowledge of all of this but it's not something I do daily
so the overly technical aspect of things end up being more confusing. I've
learned a heck of a lot more about phone systems than I ever thought I would
though and it's from taking on such projects and the good people behind
them.


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