- Talk to a service provider that provide VoIP services. - Does your PBX support SIP ? - Does your PBX also provides Topology hiding and NAT traversal , otherwise, you may need a session border controller . - Does your Service provider's softswitch has proven interworking tests with the brand of PBX you have ? - Allow PRI to SIP trunking failover and vise versa
Good luck...By the way , where is your location ? --- On Thu, 8/28/08, Tom Moore <[EMAIL PROTECTED]> wrote: > From: Tom Moore <[EMAIL PROTECTED]> > Subject: [asterisk-users] Pri to sip interfaces > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Date: Thursday, August 28, 2008, 9:06 AM > Hi guys, > What are your suggestions to people who have pbx systems > that interface with > the world over pri and want to convert them to sip > interfaces so that they > can use sip trunking? > > Tom > > > _______________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
