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

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199486#comment-199486 ] 

Joshua Colp edited comment on ASTERISK-20440 at 11/9/12 7:54 AM:
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It has not been assigned and will probably take quite awhile to get to, unless a community member decides to work on it sooner. If you want to try to speed things up you can try to find a developer who is interested to take this on or post a bounty on the asterisk-biz mailing list.
                
      was (Author: jcolp):
    It has not been assigned and will probably take quite awhile to get to, unless a community member decides to work on it sooner.
                  
> 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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