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

[Benedito Marques (JIRA)]

Benedito Marques created ASTERISK-29483:
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             Summary: 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



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