[Asterisk-Users] DUNDi in 1.2.7.1
Ludovit Koren
lk at tempest.sk
Wed May 24 08:56:06 MST 2006
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:${SECRET}@${IPADDR}/${NUMBER},nopartial
....
[00:11:43:3d:69:e6]
model = symmetric
host = lk107.tempest.sk
inkey = lk107.tempest.sk
outkey = lk40.tempest.sk
include = priv
permit = priv
qualify = yes
order = primary
and the second one:
....
priv => local1,0,IAX2,priv:${SECRET}@${IPADDR}/${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
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