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: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] 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) > > [EMAIL PROTECTED] 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 > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
