Comments (3)
Forgot to mention that when dialing in the "hang up" works fine.
Original comment by [email protected]
on 22 Sep 2012 at 1:53
from sipml5.
This is conform to RFC 3261
BYE is sent if the call is connected (200 OK)
DECLINE is sent in response to INVITE
CANCEL is send to abort pending outgoing INVITE
Original comment by [email protected]
on 22 Sep 2012 at 4:14
- Changed state: Invalid
from sipml5.
Hi,
Don't get me wrong. I am not a specialist, it is just my outgoing calls to
sip trunks were not hanging up. I think the CANCEL is sent accoringly to
RFC, but the state of the call is tsip_dialog_state_e.EARLY and it should
send HANGUP, not CANCEL.
So it hangs up also the trunk peer, not only the connection between client
and asterisk.
I commented /*&& this.o_session.b_server*/ and the dialouts are going and
hanging up fine. (I am sure that the comment is wrong but i do not know how
to do it better) at the moment it just works.
Thank you,
Cristi.
tsip_dialog.prototype.hangup = function (o_action) {
if (this.e_state == tsip_dialog_state_e.ESTABLISHED || (this.o_session
/*&& this.o_session.b_server*/ && this.e_state ==
tsip_dialog_state_e.EARLY) || this.e_type != tsip_dialog_type_e.INVITE) {
return this.fsm_act(tsip_action_type_e.HANGUP, null, o_action); //
603 Decline or BYE
}
else {
return this.fsm_act(tsip_action_type_e.CANCEL, null, o_action); //
CANCEL -> only for INVITE dialog
}
}
Original comment by [email protected]
on 27 Sep 2012 at 1:17
from sipml5.
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