[asterisk-bugs] [JIRA] Updated: (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:all-tabpanel ]

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

    Attachment: Direct Outbound.txt
                DISA outbound.txt
                Simulated Outbound DISA.txt
                Simulated Outbound Direct.txt

Attached are log files from asterisk console with debug and verbose at 5 and SIP debug on.  Four files here... two use the DISA() function to generate dialtone and receive number to dial.  Two have the number passed in directly.

Also, the two "simulated" are when I have a SLA extension (exten 111) call another asterisk extension that is not in the SLA group (exten 101). This may be easier to read as there is no authentication challenge from the trunk provider, but I provided those as well.

Through the course of testing I have found another SLA bug for which I will open another issue.
 

> 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 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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