I use asterisk 1.2.5 and h323 that comes with addons 1.2.1.
Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and
tries to make attendant transfer to person B (local SIP phone). They speak. Then
A hangs up. Call form h323 trunk doesn't get to person B.
This is what I get on CLI.
-- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663
-- Started music on hold, class 'default', on
OOH323/xxx.xxx.xxx.xxx-5381
-- Playing 'pbx-transfer' (language 'en')
-- Executing Dial("Local/303@sip-5583,2",
"SIP/303|20|wWtT") in new stack
-- Called 303
-- SIP/303-95f1 is ringing
-- Local/303@sip-5583,1 is ringing
-- SIP/303-95f1 answered Local/303@sip-5583,2
== Spawn extension (sip, 303, 1) exited non-zero on
'Local/303@sip-5583,2'
-- Playing 'beep' (language 'en')
-- Stopped music on hold on OOH323/xxx.xxx.xxx.xxx-5381
-- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663
-- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663
-- Started music on hold, class 'default', on
OOH323/xxx.xxx.xxx.xxx-5381
-- Playing 'pbx-transfer' (language 'en')
-- Executing Dial("Local/303@sip-4889,2",
"SIP/303|20|wWtT") in new stack
-- Called 303
-- SIP/303-103d is ringing
-- Local/303@sip-4889,1 is ringing
-- SIP/303-103d answered Local/303@sip-4889,2
-- Stopped music on hold on OOH323/xxx.xxx.xxx.xxx-5381
-- Playing 'beep' (language 'en')
== Spawn extension (internal, 307, 1) exited non-zero on
'Transfered/OOH323/xxx.xxx.xxx.xxx-5381<ZOMBIE>'
pbx*CLI>
Then, when I do show channels I see 0 active channels and 1 active call?!?
--
Tomislav Parcina
tparcina#lama.hr
