[asterisk-bugs] [JIRA] (ASTERISK-20440) No ringback towards SLAstation on outbound trunk call.

[dkerr (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199484#comment-199484 ] 

dkerr commented on ASTERISK-20440:
----------------------------------

Has it been assigned to a developer?  How high up in the priority queue is it.  Any estimate when it will be looked at?  This is holding up me deploying asterisk to replace an aging analog phone system.

David
                
> No ringback towards SLAstation on outbound trunk call.
> ------------------------------------------------------
>
>                 Key: ASTERISK-20440
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20440
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_meetme, Applications/SLA
>    Affects Versions: 1.8.15.0
>         Environment: Astlinux
>            Reporter: dkerr
>            Assignee: dkerr
>         Attachments: Direct Outbound.txt, DISA outbound.txt, extensions.conf, Simulated Inbound.txt, Simulated Outbound Direct.txt, Simulated Outbound DISA.txt, sip.conf, sla.conf, SLAstation log.txt
>
>
> This is similar to ASTERISK-11549 except that the problem is with calls in the other direction. If I make an outbound call from a SLA phone using the SLAStation then there is no ringtone audible to the phone that originates the call.  Both SLA stations and SLA trunks are SIP channels and for outbound I have tried both the DISA() method of connecting to a trunk (get dialtone from DISA and then enter a number to call) and directly connecting (using Dial() application instead of DISA()).
> I have turned on debug and verbose in the asterisk CLI and can observe the SIP messages and dialplan execution.  A SIP RINGING message is received by asterisk from the trunk, but the message is not passed on to the SLA station.  I have set progressinband=yes in the sip.conf and this made no difference.

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