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

dkerr (JIRA) noreply at issues.asterisk.org
Sun Sep 23 11:07:27 CDT 2012


     [ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

dkerr updated ASTERISK-20440:
-----------------------------

    Attachment: Simulated Inbound.txt

For completeness I have attached a simulated inbound call. In this instance the caller does hear a ringtone.  To test this I have a non SLA extension (101) dial 105 which in my dialplan (extensions.conf attached) simulates what would happen on an inbound trunk call... it called SLAtrunk().  This caused the two phones in the SLA group (extensions 111 and 112) to ring and ringtone is sent back to caller (101). 

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/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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