On 13/09/13 12:31, Henrik Westerberg wrote:
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?
I have no idea why you are seeing the h extension being run before the
call ends. Its not something I have ever seen happen.
Whether or not Asterisk stays in the RTP media path makes no difference
as it will always stay in the SIP signalling path and its that which
controls the call establishment and termination.
--
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