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

[brett-asterisk at worldcall.net]

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>>bounces at lists.digium.com] On Behalf Of brett-asterisk at worldcall.net
>>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
>>    
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>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,
>>    
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>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
>>
>>
>>_____________________________________
>>