Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
........
Merged revisions 376983 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376984
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
}
ast_channel_context_set(chan, endpoint->context);
- ast_channel_exten_set(chan, "s");
+ if (ast_exists_extension(NULL, endpoint->context, endpoint->name, 1, NULL)) {
+ ast_channel_exten_set(chan, endpoint->name);
+ } else {
+ ast_channel_exten_set(chan, "s");
+ }
ast_channel_priority_set(chan, 1);
ao2_unlock(endpoint);
;
; The only supported method for DTMF is RFC2833. This is always enabled on audio streams and negotiated if possible.
+; Incoming Calls
+;
+; Incoming calls will first look for the extension matching the name of the endpoint in the configured context. If
+; no such extension exists the call will automatically fall back to the "s" extension.
+
; CallerID
;
; The incoming caller id number is populated with the username of the caller and the name is populated with the full