From 12dd36c2744b255803f1dd5eda372a9a613a6992 Mon Sep 17 00:00:00 2001
From: Russell Bryant <russell@russellbryant.com>
Date: Sat, 8 Nov 2008 21:41:03 +0000
Subject: [PATCH] Merged revisions 155513 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/trunk

........
r155513 | russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines

Remove some code that is basically a no-op.  Code above this already ensures that
the buffer is terminated.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c | 6 ------
 1 file changed, 6 deletions(-)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3c922907d4..44b9fb2f28 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -19271,12 +19271,6 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
 	if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
 		return 1;
 	ast_str_set(&req.data, 0, "%s", readbuf);
-	if (res == sizeof(req.data) - 1) {
-		ast_debug(1, "Received packet exceeds buffer. Data is possibly lost\n");
-		req.data->str[sizeof(req.data) - 1] = '\0';
- 	} else
-		req.data->str[res] = '\0';
-	req.len = res;
 
 	req.socket.fd 	= sipsock;
 	req.socket.type = SIP_TRANSPORT_UDP;