[asterisk-bugs] [JIRA] (ASTERISK-29483) BYE is not send from Asterisk to another SIP address after sbc goes down

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29483?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255409#comment-255409 ] 

Asterisk Team commented on ASTERISK-29483:
------------------------------------------

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> BYE is not send from Asterisk to another SIP address after sbc goes down
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-29483
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29483
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 16.7.0
>         Environment: 2 Kamailio 5.4 as SBC, 1 Asterisk box as PBX, Microsip softphone.
>            Reporter: Benedito Marques
>
> Hello. Im trying implement failover using pjsip to make calls hangup correctly when the main SBC  goes down in the middle of the call. I've note that when the main SBC goes down in the middle of the call, Asterisk tries to send  BYE messages to IP address where they receive INVITE, and do not peform failover to another SBC. Im using DNS A record in record-rout headers send by Kamailio to Asterisk, but the failover dont accour. Exists something to do in this particular case to force failover of new transactions of the same dialog to failover (some timers, DNS question, use of sip tcp, etc)?
> Flow of the call (INVITE):
> Microsip -> Kamailio1 -> PBX
> When the "Kamailio1" goes down (BYE or some other new transaction message):
> PBX -> Kamailio1 -> Microsip
> But what I need is (BYE or some other new transaction message):
> PBX -> Kamailio2 -> Microsip



--
This message was sent by Atlassian JIRA
(v6.2#6252)