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authorKevin P. Fleming <kpfleming@digium.com>2005-09-14 02:52:11 +0000
committerKevin P. Fleming <kpfleming@digium.com>2005-09-14 02:52:11 +0000
commitf2e6ac45cac98d60966dd4c9809d8cd9b95f4ae3 (patch)
treef29976cb7dd9e63e20e0458e5bec9b60751405df /channels/chan_sip.c
parentee696945de96424f8d102ad05dd130a3dc5d145a (diff)
don't abort Asterisk load if configuration not found (issue #5115)
don't register channel type until after config has been loaded git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/chan_sip.c')
-rwxr-xr-xchannels/chan_sip.c5
1 files changed, 2 insertions, 3 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 974702999..ca9fd5fef 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12414,15 +12414,14 @@ int load_module()
ast_log(LOG_WARNING, "Unable to create I/O context\n");
}
+ reload_config(); /* Load the configuration from sip.conf */
+
/* Make sure we can register our sip channel type */
if (ast_channel_register(&sip_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type %s\n", channeltype);
return -1;
}
- if (reload_config()) /* Load the configuration from sip.conf */
- return -1;
-
/* Register all CLI functions for SIP */
ast_cli_register_multiple(my_clis, sizeof(my_clis)/ sizeof(my_clis[0]));