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

[Keith Burns]

Yep, still lineside... you can do it with SIP too. If it was going to do
MGCP, it only makes sense if it does it properly and IS aware of the
channels on the other side of the gateway (multi-chassis trunk failover
etc)


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Kevin P. Fleming
> Sent: Monday, January 24, 2005 4:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP-T Support (I got my head in an SS7
cloud)
> 
> brett-asterisk at 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.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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