Hi few weeks ago I read about redundancy (HA) of asterisk boxes using DNS, DUNDi, so I decided to give it a try.
OS FreeBSD 6.1-RELEASE, asterisk 1.2.7.1 on one peer I get: lk110*CLI> dundi show peers EID Host Model AvgTime Status 00:11:43:3d:69:e6 195.28.109.37 (S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0 unmonitored] lk110*CLI> on another: lk107*CLI> dundi show peers EID Host Model AvgTime Status 00:02:1e:f2:25:79 195.28.109.40 (S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0 unmonitored] lk107*CLI> here is the dundi.conf of the first one: .... priv => local1,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial .... [00:11:43:3d:69:e6] model = symmetric host = lk107.tempest.sk inkey = lk107.tempest.sk outkey = lk40.tempest.sk include = priv permit = priv qualify = yes order = primary and the second one: .... priv => local1,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial .... [00:02:1e:f2:25:79] model = symmetric host = lk40.tempest.sk inkey = lk40.tempest.sk outkey = lk107.tempest.sk include = priv permit = priv qualify = yes order = primary I was able to dial from phone registered on lk107 to phone registered on lk110 but no vice versa. I have read about stability problems with DUNDi and IAX2 few days ago. Is it stable at all? Am I doing something wrong? (I followed all description I found about it on www.voip-info.org). sip.conf, extensions.conf, iax.conf are the same on both servers. After reboot of the server I am not able to call from one asterisk to another and I got the following error: lk110*CLI> -- Executing NoOp("SIP/214-9fe0", "20060524-172347 ok| now were going to dundi 201") in new stack -- Executing Macro("SIP/214-9fe0", "dundi-priv|201") in new stack -- Executing Goto("SIP/214-9fe0", "201|1") in new stack -- Goto (macro-dundi-priv,201,1) -- Executing Dial("SIP/214-9fe0", "SIP/201|10|tr") in new stack May 24 17:23:47 NOTICE[561]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Dial("SIP/214-9fe0", "SIP/201|10|tr") in new stack May 24 17:23:47 NOTICE[561]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup("SIP/214-9fe0", "") in new stack == Spawn extension (local1, 201, 4) exited non-zero on 'SIP/214-9fe0' lk110*CLI> Any hints appreciated. Regards, lk _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users