Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered:
-- Accepting AUTHENTICATED call from 192.168.10.113: << Zoiper IP > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing [010...@users:1] Dial("IAX2/2200-12940", "SIP/010...@192.168.10.11") in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 << Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2...@192.168.10.77>;tag=as3e4fedb8' << 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users