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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-07-13 20:39:34 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-07-13 20:39:34 +0000
commit0e88851fbf0b7d5cf5bdfed2bf7d977d6dbcbb4c (patch)
tree629b1623b53551f2d89ab4edd1c20e8f8034ff68
parent86a49ee8618a11a78f51e2dead2562336b7175f0 (diff)
swap the G726-32 format numbers, so that IAX2 connections with prior versions of Asterisk will still work properly
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37565 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--UPGRADE.txt6
-rw-r--r--include/asterisk/frame.h8
2 files changed, 9 insertions, 5 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 32444aef7..c810f4b98 100644
--- a/UPGRADE.txt
+++ b/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:
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 709175d68..ab9cb1b8c 100644
--- a/include/asterisk/frame.h
+++ b/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) */
-#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 */