Hi I'm having trouble routing a call between two A*k servers I admin. SERVER- A: has a simple extensions set, and just needs to Dial to server B, but authenticate as part of the dial: exten => 777,1,Dial(SIP/abc-777:mypassword at someip.no-ip.info:5071/777,40,trw) exten => 777,2,Hangup So that should pass the call to the server listening on port 5071 of someip.no-ip.info, using the username of abc-777 and password of "mypassword", and pass it into extension 777 on that server. SERVER-B has a sip account defined as: [abc-777] type=friend secret = mypassword context = local host = someotherip.no-ip.info ;disallow = all ;allow = ulaw canreinvite = no nat = yes qualify = no If I run a 'show peer abc-777' then I get a peer ID through ok. But if I try to place a call, I get the following message on SERVER-A and the call disconnects. -- Executing [777 at from-sip-UK:1] Dial("SIP/ADRIANSPHONE-09dd5178", "SIP/abc-777:mypassword at someip.no-ip.info:5070/777|40|trw") in new stack [2011-04-12 15:18:03] WARNING[13926]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats -- Called abc-777:mypassword at someip.no-ip.info:5070/777 [2011-04-12 15:18:03] NOTICE[17058]: chan_sip.c:12108 handle_response_invite: Failed to authenticate on INVITE to '"Adrian Marsh" <sip:ADRIANSPHONE at 82.XXX.XXX.26>;tag=as0ff33d62' -- SIP/someip.no-ip.info:5070/777-09b16048 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [777 at from-sip-UK:2] Hangup("SIP/ADRIANSPHONE-09dd5178", "") in new stack == Spawn extension (from-sip-UK, 777, 2) exited non-zero on 'SIP/ADRIANSPHONE-09dd5178' If I turn traces on, on SERVER-B I see the line: Found peer 'abc-777' so I think the peer is authenticating ok. The context of the user looks right for accessing extension 777 on SERVER-B. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110412/2acce8b9/attachment.htm>