| Summary: | ASTERISK-09434: Calls to SLA Stations don't disconnect. | ||
| Reporter: | Mike Thomas (mthomasslo) | Labels: | |
| Date Opened: | 2007-05-13 18:12:33 | Date Closed: | 2011-06-07 14:07:25 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/SLA |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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: [macro-slaline] 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. | ||