From: David Brooks <dbrooks@digium.com>
Date: Mon, 2 Nov 2009 20:52:53 +0000 (+0000)
Subject: SIP channel name uniqueness
X-Git-Tag: 1.4.27-rc3~20
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=e3103c39a721d34edce733e2ef2313d04905bdde;p=thirdparty%2Fasterisk.git

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f5a45b2a5f..7a53c106f3 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -398,6 +398,8 @@ static const struct  cfsip_methods {
 	{ 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
@@ -4207,7 +4209,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 
 		ast_mutex_unlock(&i->lock);
 		/* 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) {