Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.84
diff -u -r1.84 rtp.c
--- rtp.c	7 Aug 2004 14:22:09 -0000	1.84
+++ rtp.c	25 Aug 2004 08:51:35 -0000
@@ -587,7 +587,7 @@
   {{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
   {{1, AST_FORMAT_LPC10}, "audio", "LPC"},
   {{1, AST_FORMAT_G729A}, "audio", "G729"},
-  {{1, AST_FORMAT_SPEEX}, "audio", "SPEEX"},
+  {{1, AST_FORMAT_SPEEX}, "audio", "speex"},
   {{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
   {{0, AST_RTP_DTMF}, "audio", "telephone-event"},
   {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
@@ -698,6 +698,7 @@
   }
 }
 
+/* We must ALWAYS look up from our own RTP PT table when receiving RTP */
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
 {
   struct rtpPayloadType result;
@@ -706,11 +707,7 @@
     result.isAstFormat = result.code = 0;
     return result; /* bogus payload type */
   }
-  /* Start with the negotiated codecs */
-  result = rtp->current_RTP_PT[pt];
-  /* If it doesn't exist, check our static RTP type list, just in case */
-  if (!result.code) 
-  	result = static_RTP_PT[pt];
+  result = static_RTP_PT[pt];
   return result;
 }