This is an upgrade from a previous system. The old one didn't handle PRI, so they had analog phone lines as trunks. Management won't invest the money right now to get a PRI circuit. Any suggestions?
> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith > Sent: Monday, July 19, 2004 4:59 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] PSTN gateway implementation? > > > -I have a TE405P board and only one T1 worth of phone lines (24) > > connected to it using an Adtran TA750 channel bank. > > Any particular reason against using PRI from your telco? > > > Is Asterisk capable of handling multiple incoming VoIP calls arriving > > from the same source (IP) or do I need to get something else to take the > > incoming traffic and pass it on to Asterisk? (I've read about using SER > > as a SIP proxy, but it's not clear to me wheather I need it or not). Can > > I use the OpenH.323 module to take care of the incoming VoIP traffic? > > Asterisk can handle multiple calls from the same IP without any worry. > Your > main worry is the lack of real billing since you're terminating to analog > PSTN instead of using PRI -- you have no way of actually knowing if the > call > was answered or not, so he'll be billed on every call. I doubt you want > to > try and work with callprogress=yes. > > -A. > _______________________________________________ > 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