Version 0.1.0 from FTP
authorMark Spencer <markster@digium.com>
Sat, 4 Dec 1999 22:05:35 +0000 (22:05 +0000)
committerMark Spencer <markster@digium.com>
Sat, 4 Dec 1999 22:05:35 +0000 (22:05 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85 65c4cc65-6c06-0410-ace0-fbb531ad65f3

README [new file with mode: 0755]
codecs/gsm_slin_ex.h [new file with mode: 0755]

diff --git a/README b/README
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--- /dev/null
+++ b/README
@@ -0,0 +1,27 @@
+The Asterisk Open Source PBX
+by Mark Spencer <markster@linux-support.net>
+Copyright (C) 1999, Linux Support Services, LLC and Adtran, Inc.
+================================================================
+* WHAT IS ASTERISK
+  Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
+sense, middleware between Internet and telephony channels on the bottom,
+and Internet and telephony applications at the top.  For more information
+on the project itself, please visit the Asterisk home page at:
+
+           http://www.asteriskpbx.com
+
+
+* REQUIRED COMPONENTS
+
+== Linux ==
+  Currently, the Asterisk Open Source PBX is only known to run on the
+Linux OS, although it may be portable to other UNIX-like operating systems
+as well.
+
+== libaudiofile ==
+
+  If you want to use format_wav module, then you need a very recent
+version of libaudiofile (at least version 0.2.0, or you can apply the
+following patch to version 0.1.9):
+
+
diff --git a/codecs/gsm_slin_ex.h b/codecs/gsm_slin_ex.h
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+/*
+  * 8-bit raw data
+  *
+  * Source: gsm.example
+  *
+  * Copyright (C) 1999, Mark Spencer and Linux Support Services
+  *
+  * Distributed under the terms of the GNU General Public License
+  *
+  */
+
+static unsigned char gsm_slin_ex[] = {
+0xda, 0xa6, 0xac, 0x2d, 0xa3, 0x50, 000, 0x49, 0x24, 0x92, 
+0x49, 0x24, 0x50, 0x40, 0x49, 0x24, 0x92, 0x37, 0x24, 0x52, 
+000, 0x49, 0x24, 0x92, 0x47, 0x24, 0x50, 0x80, 0x46, 0xe3, 
+0x6d, 0xb8, 0xdc };