[asterisk-ha-clustering] Trouble with Real Time

Edgar Guadamuz eguadamuz at gmail.com
Fri May 9 13:17:22 CDT 2008


Hi dudes, I have two asterisk boxes using realtime:

Node1: 192.168.1.3 (Asterisk server)

Node2: 192.168.1.5 (Asterisk and MySQL server)


Node1 connects to Node2 Database for retrieve data. Both nodes are
under Heartbeat monitoring, and that works fine. The issue is: I
registered a phone 103 to Node1 and phone 102 to Node 2. When I make a
call  from 103 to 102 Node 1 gets the info about extension 102 I don't
know how:

    -- Executing Dial("SIP/103-081a7788", "SIP/102")
    -- SIP Seeding peer from astdb: '102' at 102 at 192.168.1.5:5061 for 1800
    -- Called 102
May  9 18:37:45 WARNING[6730]: channel.c:2781
ast_channel_make_compatible: No path to translate from
SIP/102-081ad4c8(256) to SIP/103-081a7788(1024)
    -- SIP/102-081ad4c8 is ringing
  == Spawn extension (default, 102, 1) exited non-zero on 'SIP/103-081a7788'


And the "sip show peers" shows both extensions. However, in the other
way, when I try to make a call from 102 to 103 the "Seeding peer from
astdb" doesn't appears and the call fails.


    -- Executing Dial("SIP/102-081be5d0", "SIP/103")
May  9 12:16:01 NOTICE[20487]: app_dial.c:1076 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 Goto("SIP/102-081be5d0", "lookupdundi|103|1")
    -- Goto (lookupdundi,103,1)
May  9 12:16:01 WARNING[20487]: pbx.c:2377 __ast_pbx_run: Channel
'SIP/102-081be5d0' sent into invalid extension '103' in context
'lookupdundi', but no invalid handler


I've also try with dundi but it didn't work either, as you can notice.
Any suggestion will be welcome.



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