Re: [Asterisk-Users] SIP-T Support (I got my head in an SS7 cloud)

Mon, 24 Jan 2005 20:59:25 -0800

I agree with your comparison if *,and the like as line side feature serves, however I don't particularly agree with your class 5 definition. Perhaps you meant class 4? Class 5 is typically the line side termination switch. IE your typical POTS line is attached to a Class 5 switch of some sort. Class 4 switches don't have "features" and don't typically have networks (it's the NNI not the UNI).

All that being said, look in the PSTN. SS7 connection connect Class 5 to Class 5, Class 4 to Class 4 and Class 4 to Class 5. Therefore, there is most likely a place to put SS7 into a box with line side features..

Now.. that's not necessarily what I want to do with it. Here's the deal.

1. I have invested in a fancy, expensive SS7 gateway. A Sonus Network. I have extra A links and point codes. I believe my equipment supports SIP-T, but I haven't experimented much with these features yet.
2. I have a SIP platform that should be able to parse SIP headers. Including extended SIP headers found in SIP-T (or so I'm hoping)
3. MGCP allows you to address a remote DS0.
4. Asterisk can connect a SIP to a MGCP call.
<stretch>
5. Both MGCP and SIP support reinvite methods.
6. Asterisk might be able to bridge between MGCP and SIP-T?
</stretch>

Of course it'd take some doing. I could leave out the reinvite stuff if I could get everything else to work.

Like I said, got my head in the clouds.. Perhaps I should just focus on hardware that can act as a TDM endpoint for a Sonus PSX (such as the audiocodes system).
-Brett



Keith Burns wrote:

I think of *, Broadworks, Vocaldata, Sylantro as "line side feature
servers", and SS7 signaling with say IMTs/PRIs more for the class5
network side soft-switch (NexVerse, SONUS etc).

Typically they handle the LERG, complex translations etc and do it quite
well (although typically they take in native A-links for SS7 or some
degree of the "SS7-o-IP" standards).

I'm not sure I would want a line side feature server trying to be all
things to all people... kinda gets like Cisco IOS Enterprise :-o





-----Original Message-----
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, January 24, 2005 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP-T Support (I got my head in an SS7


cloud)


Hey All,
I'm just daydreaming here.. but what's the status of SIP-T in


Asterisk?


I haven't been able to find a whole lot of info on SIP-T but seems


like


just an extension of SIP. Right?

Now if I had a PSTN Gateway (that is a SS7 gateway) that supported
SIP-T, could I signal * with SIP-T from it and have asterisk utilize
MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am I


missing


here.. ??

Hmm, but outbound calls would be more complicated I think.. Let see,


SIP


user dials a number, we'll eventually  place a dial out on the MGCP
line, but we need to first send a few SIP-T messages to find out where
to put it..

Just swiming around in it here.. Any thoughts? It seems to me that you
MUST use something like MGCP or H.248 to connect the call to the PSTN
(media gateway) since the specific DS0 to be utilized will be included
in the ISUP messages..

-Brett


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