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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-07-13 20:39:34 +00:00
parent 4376af0080
commit 4492cbac8d
2 changed files with 9 additions and 5 deletions
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6
UPGRADE.txt
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@@ -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:

8
include/asterisk/frame.h
View File

@@ -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) */
#define AST_FORMAT_G726 (1 << 4)
/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
#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) */
#define AST_FORMAT_G726_AAL2 (1 << 11)
/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
#define AST_FORMAT_G726 (1 << 11)
/*! Maximum audio format */
#define AST_FORMAT_MAX_AUDIO (1 << 15)
/*! Maximum audio mask */