[asterisk-commits] russell: branch 1.4 r54204 -
/branches/1.4/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Feb 13 12:42:01 MST 2007
Author: russell
Date: Tue Feb 13 13:42:00 2007
New Revision: 54204
URL: http://svn.digium.com/view/asterisk?view=rev&rev=54204
Log:
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.
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=54204&r1=54203&r2=54204
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Feb 13 13:42:00 2007
@@ -16510,6 +16510,7 @@
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;
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