asterisk users - Jan 2005 - 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

brett-asterisk@worldcall.net wrote:
> 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..
No, you can just do what you are doing now, and use SIP to talk to your 
gateway. The SIP "user" (Asterisk) has no concept of how many channels
exist on the TDM side, or their arrangement, or anything like that.

If Asterisk could be an MGCP gateway controller (whatever the right term 
for that is) it's possible that it could control MGCP gateways directly, 
but it would still need to speak some sort of signaling with the PSTN to 
setup/teardown the calls.

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@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of brett-asterisk@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
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users