[asterisk-commits] file: branch oej/earlyrtpfix r53101 - in /team/oej/earlyrtpfix: ./ channels/ ...

[asterisk-commits at lists.digium.com]

Author: file
Date: Thu Feb  1 16:08:26 2007
New Revision: 53101

URL: http://svn.digium.com/view/asterisk?view=rev&rev=53101
Log:
Merged revisions 53097,53099 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines

Merged revisions 53095 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines

Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) 

........

................
r53099 | file | 2007-02-01 16:04:58 -0600 (Thu, 01 Feb 2007) | 2 lines

Huh... fix the berkeley DB to compile here as well, but it apparently required both dev mode and no optimizations to creep up.

................

Modified:
    team/oej/earlyrtpfix/   (props changed)
    team/oej/earlyrtpfix/channels/chan_sip.c
    team/oej/earlyrtpfix/main/db1-ast/hash/hash.c

Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Feb  1 16:08:26 2007
@@ -1,1 +1,1 @@
-/branches/1.4:1-53095
+/branches/1.4:1-53100

Modified: team/oej/earlyrtpfix/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_sip.c?view=diff&rev=53101&r1=53100&r2=53101
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_sip.c (original)
+++ team/oej/earlyrtpfix/channels/chan_sip.c Thu Feb  1 16:08:26 2007
@@ -559,7 +559,6 @@
 
 /*! \brief Codecs that we support by default: */
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
 
 /* Object counters */
 static int suserobjs = 0;                /*!< Static users */
@@ -944,6 +943,7 @@
 	int peercapability;			/*!< Supported peer capability */
 	int prefcodec;				/*!< Preferred codec (outbound only) */
 	int noncodeccapability;			/*!< DTMF RFC2833 telephony-event */
+	int jointnoncodeccapability;            /*!< Joint Non codec capability */
 	int redircodecs;			/*!< Redirect codecs */
 	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
 	struct t38properties t38;		/*!< T38 settings */
@@ -5120,7 +5120,7 @@
 
 	newjointcapability = p->capability & (peercapability | vpeercapability);
 	newpeercapability = (peercapability | vpeercapability);
-	newnoncodeccapability = noncodeccapability & peernoncodeccapability;
+	newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
 		
 		
 	if (debug) {
@@ -5134,7 +5134,7 @@
 			    ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
 
 		ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
-			    ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
+			    ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
 			    ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
 			    ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
 	}
@@ -5153,9 +5153,9 @@
 
 	/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
 		they are acceptable */
-	p->jointcapability = newjointcapability;	/* Our joint codec profile for this call */
-	p->peercapability = newpeercapability;		/* The other sides capability in latest offer */
-	p->noncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
+	p->jointcapability = newjointcapability;	        /* Our joint codec profile for this call */
+	p->peercapability = newpeercapability;		        /* The other sides capability in latest offer */
+	p->jointnoncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
 
 	ast_rtp_pt_copy(p->rtp, newaudiortp);
 	if (p->vrtp)
@@ -6301,7 +6301,7 @@
 
 	/* Now add DTMF RFC2833 telephony-event as a codec */
 	for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
-		if (!(p->noncodeccapability & x))
+		if (!(p->jointnoncodeccapability & x))
 			continue;
 
 		add_noncodec_to_sdp(p, x, 8000,

Modified: team/oej/earlyrtpfix/main/db1-ast/hash/hash.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/main/db1-ast/hash/hash.c?view=diff&rev=53101&r1=53100&r2=53101
==============================================================================
--- team/oej/earlyrtpfix/main/db1-ast/hash/hash.c (original)
+++ team/oej/earlyrtpfix/main/db1-ast/hash/hash.c Thu Feb  1 16:08:26 2007
@@ -728,7 +728,7 @@
 	u_int32_t flag;
 {
 	register u_int32_t bucket;
-	register BUFHEAD *bufp;
+	register BUFHEAD *bufp = NULL;
 	HTAB *hashp;
 	u_int16_t *bp, ndx;