[asterisk-bugs] [JIRA] (ASTERISK-21663) Realtime TCP endpoints lose registration after "sip reload" & "core reload"
Michael L. Young (JIRA)
noreply at issues.asterisk.org
Sun Apr 21 08:47:01 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21663?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205505#comment-205505 ]
Michael L. Young commented on ASTERISK-21663:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
What is the "transport" setting for the peer in your realtime table? The default will be UDP if the transport is not specified for that peer and is only specified in the global settings.
Also, is "rtcachefriends" turned on?
> 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
> Affects Versions: 1.8.21.0, 11.3.0
> Reporter: Dinesh Ramjuttun
> Severity: Minor
>
> 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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