From: drowe67 Date: Tue, 29 May 2012 23:47:17 +0000 (+0000) Subject: change for 2400 and README typos, thanks Ed & Hisaharu X-Git-Url: http://git.whiteaudio.com/gitweb/?a=commitdiff_plain;h=eb97c26a990884591262371d3ad8763b2754733d;p=freetel-svn-tracking.git 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 1dcbdf25..a2d3b512 100644 --- 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 5c0efada..5d477f65 100644 --- 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; }