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

Michael L. Young (JIRA) noreply at issues.asterisk.org
Sat Nov 10 22:04:21 CST 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199538#comment-199538 ] 

Michael L. Young commented on ASTERISK-20440:
---------------------------------------------

Same comment as in the other issue.  No need to include the issue number in the comment.

Thanks
                
> 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: asterisk-1.8-bugid20440.patch, 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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