From 83856d4683d5409a76f62bd1917cc5f049223651 Mon Sep 17 00:00:00 2001 From: Russell Bryant Date: Tue, 13 Feb 2007 21:57:31 +0000 Subject: [PATCH] Merged revisions 54204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines If we fail to create the SIP socket, then return -1 from reload_config() so that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get spammed with error messages every time chan_sip tries to send a message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 1 + 1 file changed, 1 insertion(+) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index fd153f899a..d4713f488b 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -16882,6 +16882,7 @@ static int reload_config(enum channelreloadreason reason) sipsock = socket(AF_INET, SOCK_DGRAM, 0); if (sipsock < 0) { ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno)); + return -1; } else { /* Allow SIP clients on the same host to access us: */ const int reuseFlag = 1;