[asterisk-bugs] [JIRA] (ASTERISK-21663) [patch] Realtime TCP endpoints lose registration after "sip reload" & "core reload"

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21663?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242263#comment-242263 ] 

Michael L. Young commented on ASTERISK-21663:
---------------------------------------------

This issue was closed 5 years ago with no response from the original reporter.  If this issue is still present in the latest supported version of Asterisk, I would recommend that you open up a new report.  You can always reference this issue.

> [patch] Realtime TCP endpoints lose registration after "sip reload" &  "core reload"
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21663
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21663
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/DatabaseSupport
>    Affects Versions: 1.8.21.0, 11.3.0
>            Reporter: Dinesh Ramjuttun
>            Assignee: Michael L. Young
>            Severity: Minor
>              Labels: patch
>         Attachments: 21663.diff, asterisk-21663-outbound-default-transport-fix_11.diff, asterisk-21663-outbound-default-transport-fix_1.8.diff
>
>
> The scenario is as follows:
> - TCP endpoints are being used.
> - transport is set to "udp,tcp" in sip.conf (transport=udp,tcp)
> I have tested with both realtime configuration and flat peer configuration in sip.conf
> After a "sip reload", a realtime TCP peer loses its registration. With qualify=yes set, the TCP peer becomes "unreachable" to asterisk. When that TCP peer is called, sip invites are retransmitted unsuccessfully before giving up. Extension to extension calls cannot go through. Only way to fix this is either by restarting Asterisk or waiting for the peers to re-register again.
> If peer setting is static in sip.conf, the TCP endpoint does not lose its registration.
> I have compared the "sip show peer [peer]" in both cases after a "sip reload" or "core reload". With realtime peers, "sip show peer [peer]" shows primary transport as UDP while "sip show peer [peer]" with static peer in sip.conf ,primary transport is showed as TCP.



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