[asterisk-commits] pkiefer: trunk r376820 - /trunk/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: pkiefer
Date: Thu Nov 29 10:44:42 2012
New Revision: 376820

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=376820
Log:
Fix chan_sip websocket payload handling

Websocket by default doesn't return an ast_str for the payload received. When 
converting it to an ast_str on chan_sip the last character was being omitted, 
because ast_str functions expects that the given length includes the trailing 
0x00. payload_len only has the actual string length without counting the 
trailing zero.

For most cases this passed unnoticed as most of SIP messages ends with \r\n.

(closes issue ASTERISK-20745)
Reported by: Iñaki Baz Castillo
Review: https://reviewboard.asterisk.org/r/2219/

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=376820&r1=376819&r2=376820
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov 29 10:44:42 2012
@@ -2635,7 +2635,7 @@
 		if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
 			struct sip_request req = { 0, };
 
-			if (!(req.data = ast_str_create(payload_len))) {
+			if (!(req.data = ast_str_create(payload_len + 1))) {
 				goto end;
 			}