Bug#340574: asterisk: SIP error 400 on outgoing calls

[Elmar Haneke]

Mike Fedyk schrieb:
> Elmar Haneke wrote:
> 
>> The relevant patch should be the change from rev 1.913 to 1.914:
>>
>> 4270c4270
>> <               ast_build_string(a_buf, a_size, "a=fmtp:%d 
>> annexb=no",rtp_code);
>> ---
>>  
>>
>>>               ast_build_string(a_buf, a_size, "a=fmtp:%d 
>>> annexb=no\r\n",rtp_code);
>>>     
> 
> Please post the patch in diff -u format.  Thanks.
> 
> 
> 

-- 
Mit freundlichen Grüßen

Elmar Haneke

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|Dr. Elmar Haneke                   Tel: +49-2241-39749-0 |
|Haneke Software                    Fax: +49-2241-39749-30|
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-------------- next part --------------
--- chan_sip.c	16 Nov 2005 17:46:59 -0000	1.913
+++ chan_sip.c	17 Nov 2005 20:25:40 -0000	1.914
@@ -50,7 +50,7 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision: 1.913 $")
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 1.914 $")
 
 #include "asterisk/lock.h"
 #include "asterisk/channel.h"
@@ -4267,7 +4267,7 @@
 			 sample_rate);
 	if (codec == AST_FORMAT_G729A)
 		/* Indicate that we don't support VAD (G.729 annex B) */
-		ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no", rtp_code);
+		ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
 }
 
 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,