[asterisk-commits] dbrooks: branch 1.6.2 r226978 - /branches/1.6.2/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dbrooks
Date: Mon Nov  2 15:06:20 2009
New Revision: 226978

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=226978
Log:
SIP channel name uniqueness

SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.

(closes issue #15152)
Reported by: palbrecht

Review: https://reviewboard.asterisk.org/r/420/

Modified:
    branches/1.6.2/channels/chan_sip.c

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=226978&r1=226977&r2=226978
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Mon Nov  2 15:06:20 2009
@@ -850,6 +850,8 @@
 	{ SIP_PING,	 NO_RTP, "PING", 	CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
 };
 
+static unsigned int chan_idx;
+
 /*!  Define SIP option tags, used in Require: and Supported: headers 
  	We need to be aware of these properties in the phones to use 
 	the replace: header. We should not do that without knowing
@@ -6619,7 +6621,7 @@
 
 		sip_pvt_unlock(i);
 		/* Don't hold a sip pvt lock while we allocate a channel */
-		tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i);
+		tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, ast_atomic_fetchadd_int((int *)&chan_idx, +1));
 
 	}
 	if (!tmp) {