Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always
over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten => h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten => _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten => _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =>
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=failed&dialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
Regards,
Henrik
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