swap the G726-32 format numbers, so that IAX2 connections with prior versions of Asterisk will still work properly

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37565 65c4cc65-6c06-0410-ace0-fbb531ad65f3

UPGRADE.txt

@@ -296,7 +296,11 @@ The G726-32 codec:
   be able to continue to use these devices with this version of Asterisk and the
   G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
   to sip.conf, so that Asterisk can use the packing order expected by the device (even
-  though it requested a different order).
+  though it requested a different order). In addition, the internal format number for
+  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
+  result of this is that this version of Asterisk will be able to interoperate over
+  IAX2 with older versions of Asterisk, as long as this version is told to allow
+  'g726aal2' instead of 'g726' as the codec for the call.
 
 Installation:
 

include/asterisk/frame.h

@@ -214,8 +214,8 @@ extern struct ast_frame ast_null_frame;
 #define AST_FORMAT_ULAW		(1 << 2)
 /*! Raw A-law data (G.711) */
 #define AST_FORMAT_ALAW		(1 << 3)
-/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
+/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
-#define AST_FORMAT_G726		(1 << 4)
+#define AST_FORMAT_G726_AAL2	(1 << 4)
 /*! ADPCM (IMA) */
 #define AST_FORMAT_ADPCM	(1 << 5)
 /*! Raw 16-bit Signed Linear (8000 Hz) PCM */
@@ -228,8 +228,8 @@ extern struct ast_frame ast_null_frame;
 #define AST_FORMAT_SPEEX	(1 << 9)
 /*! iLBC Free Compression */
 #define AST_FORMAT_ILBC		(1 << 10)
-/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
+/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
-#define AST_FORMAT_G726_AAL2	(1 << 11)
+#define AST_FORMAT_G726		(1 << 11)
 /*! Maximum audio format */
 #define AST_FORMAT_MAX_AUDIO	(1 << 15)
 /*! Maximum audio mask */