Shaun Wingrin wrote: > The setup is as follows: SIP phone registers via international link > to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 > via Zaptel Channels need to be hairpinned from Box 1 to 2. How is > sip.conf configured on Box 1 and 2 so that we don't get an error: > "Failed to authenticate user" when 1's extensions.conf uses SIP to > dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP > phone registering at 1 directly to 2 without first passing through 2? > > Tx > > Shaun > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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 This happens through a sip re-invite, the problem you seem to be having is that box 1 is not authenticated to send calls to box 2.
Anthony /"Everything should be as simple as possible, but no simpler" - Albert Einstien/ _______________________________________________ -- 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
