typos

git-svn-id: https://svn.code.sf.net/p/freetel/code@361 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/codec2-dev/asterisk/README b/codec2-dev/asterisk/README
index 70e58486932d54b415341b1009a93cb9d7fb5163..1dcbdf25e603ffbd73853947267bdccd9759fcef 100644 (file)
--- a/codec2-dev/asterisk/README
+++ b/codec2-dev/asterisk/README
@@ -10,10 +10,10 @@ Codec 2 is used to trunk calls between two Asterisk boxes:
 
 The two SIP phones are configiured for mulaw.
 
-Buildling
+Building
 ---------
 
-Asterisk must be pacthed so that the core understand Codec 2 frames.
+Asterisk must be patched so that the core understand Codec 2 frames.
 
 1/ First install Codec 2: