Hi guys,
I'm testing the new gxw-4024 appliance but have a problem with attended
transfer, it works but after that the phone transfered the call, it
results "busy" for 60 seconds.
In my scenario the phone connected to 4024 (phone B) receive a call from
another sip client logged on asterisk server (phone A), it put it on
hold by pressing "R" (flash button) and dial another sip client also
logged on my asterisk (phone C). This one speak with B and accepts the
call. At this point, B hangs up by putting down the handset and let A
speaks with C.
I registered the port1 on asterisk server configured as follow (sip.conf
and extensions.conf ):
[207]
type = friend
username = password
host = dynamic
nat = never
port = 5060
context = per_tutti
secret = 207
dtmfmode = inband
canreinvite = yes
language = it
canreinvite = yes
mailbox = 207
qualify = yes
callerid = Test <207>
[local]
exten => _[24]XX,1,Macro(exten,${EXTEN})
exten => _[24]XX,2,HangUp
[macro-exten]
exten => s,1,Dial(${ARG1})
exten => s,2,GoTo(s-${DIALSTATUS},1)
exten => s-BUSY,1,Busy()
exten => s-BUSY,2,HangUp
exten => s-NOANSWER,1,Congestion()
exten => s-NOANSWER,2,HangUp
exten => s-CONGESTION,1,Congestion()
exten => s-CONGESTION,2,HangUp
exten => s-CANCEL,1,Congestion()
exten => s-CANCEL,2,HangUp
As anyone tried similar scenario?
Thanks all
Giordano Grandis
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