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

[Keith Burns]

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
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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