change for 2400 and README typos, thanks Ed & Hisaharu

git-svn-id: https://svn.code.sf.net/p/freetel/code@519 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/codec2-dev/asterisk/README b/codec2-dev/asterisk/README
index 1dcbdf25e603ffbd73853947267bdccd9759fcef..a2d3b512370b05564fe308888203e055110f2895 100644 (file)
--- a/codec2-dev/asterisk/README
+++ b/codec2-dev/asterisk/README
@@ -8,7 +8,7 @@ Codec 2 is used to trunk calls between two Asterisk boxes:
 
     A - SIP phone - Asterisk A - Codec2 - Asterisk B - SIP Phone - B
 
-The two SIP phones are configiured for mulaw.
+The two SIP phones are configured for mulaw.
 
 Building
 ---------
@@ -27,8 +27,8 @@ Asterisk must be patched so that the core understand Codec 2 frames.
 
     david@cool:~$ tar xvzf asterisk-1.8.9.0.tar.gz
     david@cool:~/asterisk-1.8.9.0$ patch -p4 < ~/codec2-dev/asterisk/asterisk-codec2.patch
-    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec2_codec2.c .
-    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h .
+    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec_codec2.c .
+    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h ./codecs
     david@cool:~/asterisk-1.8.9.0$ ./configure && make ASTLDFLAGS=-lcodec2
     david@cool:~/asterisk-1.8.9.0$ sudo make install
     david@cool:~/asterisk-1.8.9.0$ sudo make samples

diff --git a/codec2-dev/asterisk/codec_codec2.c b/codec2-dev/asterisk/codec_codec2.c
index 5c0efada82589c42d4a829c4e638c6a018c410f6..5d477f65795a7dc3b84ff9659f48f55f06ddeb99 100644 (file)
--- a/codec2-dev/asterisk/codec_codec2.c
+++ b/codec2-dev/asterisk/codec_codec2.c
@@ -48,7 +48,7 @@ static int codec2_new(struct ast_trans_pvt *pvt)
 {
     struct codec2_translator_pvt *tmp = pvt->pvt;
 
-    tmp->codec2 = codec2_create(CODEC2_MODE_2500);
+    tmp->codec2 = codec2_create(CODEC2_MODE_2400);
        
     return 0;
 }