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authorMark Michelson <mmichelson@digium.com>2007-11-19 23:24:35 +0000
committerMark Michelson <mmichelson@digium.com>2007-11-19 23:24:35 +0000
commitfb3b4f493796c4d1412d3acabd93703e9c092033 (patch)
tree73485470b8cf7b3f3fe3a05fa89b10408995c5ea
parent3cda8388b93fdf0f7f34990a58fb803a7dba35b3 (diff)
Changed the "busy-level" option in sip.conf to "busylevel" to be more parallel
with the SIPPEER() argument of the same name. The deprecation procedure is not being used here since this is a trunk-only option. (closes issue #11307, reported by pj, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r--CHANGES2
-rw-r--r--channels/chan_sip.c2
2 files changed, 2 insertions, 2 deletions
diff --git a/CHANGES b/CHANGES
index 0f0aa88e6..e92391804 100644
--- a/CHANGES
+++ b/CHANGES
@@ -68,7 +68,7 @@ SIP changes
since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
* The "localmask" setting was removed in version 1.2 and the reminder about it
being removed is now also removed.
- * A new option "busy-level" for setting a level of calls where asterisk reports
+ * A new option "busylevel" for setting a level of calls where asterisk reports
a device as busy, to separate it from call-limit. This value is also added
to the SIP_PEER dialplan function.
* A new realtime family called "sipregs" is now supported to store SIP registration
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6df7e1d0b..3c2d73b36 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17811,7 +17811,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
peer->call_limit = atoi(v->value);
if (peer->call_limit < 0)
peer->call_limit = 0;
- } else if (!strcasecmp(v->name, "busy-level")) {
+ } else if (!strcasecmp(v->name, "busylevel")) {
peer->busy_level = atoi(v->value);
if (peer->busy_level < 0)
peer->busy_level = 0;