[asterisk-users] Forking using Openser And Asterisk

[Aadilkhan Maniyar]

Hi All,
 
I am stuck with an issue in the Openser+Asterisk Forking. 
 
In this solution we are using Openser as the Registrar. Hence it will
store all the contact bindings along with the q values for a given user,
say ua1. The current setup is such that the INVITEs are sent to Asterisk
by Openser and Asterisk sends out the INVITE.
 
Now if ua1 is registered with two different contacts having different q
values and i make a call from ua2 to ua1.
Openser will recieve the INVITE check for the multiple contacts of ua1
in the database. and send out an INVITE for the first contact. On
recieving a 486 busy it sends out an INVITE to the second contact. This
is where the problems lies.
 
Openser is sending Asterisk the second INVITE but none of the actions
specified in the Dialplan (extensions.conf) of Asterisk seem to be
executing on reciept of the second INVITE.
 
My extensions.conf looks like this:
    exten => _.,1,NoOp(Incoming Call ${EXTEN}@${SIPDOMAIN})
    exten => _.,2,Dial(SIP/${EXTEN})
    exten => _.,3,HangUp()
    exten => h,4,HangUp()
 
 
This is the ouput at the asterisk cli:
 
    -- Executing [ua1 at call:1] NoOp("SIP/ua2-0921a250", "Incoming Call
ua1 at 10.0.16.30") in new stack
    -- Executing [ua1 at call:2] Dial("SIP/ua2-0921a250", "SIP/ua1") in new
stack
    -- Called ua1
[Apr  3 17:02:41] NOTICE[20198]: chan_sip.c:2918 auto_congest:
Auto-congesting SIP/ua1-0921f080
    -- SIP/ua1-0921f080 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [ua1 at call:3] Hangup("SIP/ua2-0921a250", "") in new
stack
  == Spawn extension (call, ua1, 3) exited non-zero on
'SIP/ua2-0921a250'
    -- Executing [h at call:1] NoOp("SIP/ua2-0921a250", "Incoming Call from
house extension  for h at 10.0.16.30") in new stack
    -- Executing [h at call:3] Dial("SIP/ua2-0921a250", "SIP/h") in new
stack
[Apr  3 17:02:51] WARNING[21381]: chan_sip.c:2898 create_addr: No such
host: h
[Apr  3 17:02:51] WARNING[21381]: app_dial.c:1191 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 [h at call:3] Hangup("SIP/ua2-0921a250", "") in new stack
  == Spawn extension (call, h, 3) exited non-zero on 'SIP/ua2-0921a250'
If anyone has any inputs on this I would appreciate it..
 
Thanks & Regards,
Aadil
 
 
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