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

[Kevin P. Fleming]

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.