[asterisk-commits] mmichelson: branch group/CCSS r227047 - in /team/group/CCSS: ./ channels/ main/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Nov 2 15:38:24 CST 2009


Author: mmichelson
Date: Mon Nov  2 15:38:20 2009
New Revision: 227047

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=227047
Log:
Resolve conflict and reset automerge.

I'm very disappointed in svn for not being able to resolve
this one on its own. Seriously.


Modified:
    team/group/CCSS/   (props changed)
    team/group/CCSS/channels/chan_sip.c
    team/group/CCSS/main/http.c

Propchange: team/group/CCSS/
------------------------------------------------------------------------------
    automerge = *

Propchange: team/group/CCSS/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.

Propchange: team/group/CCSS/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Nov  2 15:38:20 2009
@@ -1,1 +1,1 @@
-/trunk:1-226965
+/trunk:1-227046

Modified: team/group/CCSS/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/channels/chan_sip.c?view=diff&rev=227047&r1=227046&r2=227047
==============================================================================
--- team/group/CCSS/channels/chan_sip.c (original)
+++ team/group/CCSS/channels/chan_sip.c Mon Nov  2 15:38:20 2009
@@ -930,6 +930,8 @@
 	{ 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
@@ -6946,7 +6948,7 @@
 
 		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, linkedid, 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, linkedid, i->amaflags, "SIP/%s-%08x", my_name, ast_atomic_fetchadd_int((int *)&chan_idx, +1));
 	}
 	if (!tmp) {
 		ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");

Modified: team/group/CCSS/main/http.c
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/main/http.c?view=diff&rev=227047&r1=227046&r2=227047
==============================================================================
--- team/group/CCSS/main/http.c (original)
+++ team/group/CCSS/main/http.c Mon Nov  2 15:38:20 2009
@@ -24,6 +24,8 @@
  *
  * This program implements a tiny http server
  * and was inspired by micro-httpd by Jef Poskanzer
+ *
+ * \extref GMime http://spruce.sourceforge.net/gmime/
  *
  * \ref AstHTTP - AMI over the http protocol
  */




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