Summary:ASTERISK-09434: Calls to SLA Stations don't disconnect.
Reporter:Mike Thomas (mthomasslo)Labels:
Date Opened:2007-05-13 18:12:33Date Closed:2011-06-07 14:07:25
Versions:Frequency of
Description:When a incoming call on a SIP trunk is answered by a SLAStation the call proceeds as expected.  However if the SIP end point hangs-up before the caller, rather than disconnecting, the caller is connected to the next step in the dial plan.  For example an outside caller calls into a SLA, it rings a phone (or group of phones) and it's answered by one of them.  At the conclusion of the call the asterisk user "hangs-up" their SIP endpoint, at this point the outside caller advances to the next step within the same context in the dial plan in extensions.conf which is typically voicemail.
Comments:By: Russell Bryant (russell) 2007-05-14 12:44:24

You should be using the SLATRUNK_STATUS variable to determine whether you should go into voicemail or not.  Here is a macro taken from the SLA with Voicemail example in sla.pdf:

exten => s,1,SLATrunk(${ARG1})
exten => s,n,Goto(s-${SLATRUNK_STATUS}|1)
exten => s-FAILURE,1,Voicemail(1234|u)
exten => s-UNANSWERED,1,Voicemail(1234|u)

In the case that you described, SLATRUNK_STATUS would be set to "SUCCESS", so it wouldn't hit voicemail, it would just be hung up.