[asterisk-commits] wdoekes: branch 1.8 r375074 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: wdoekes
Date: Tue Oct 16 14:21:39 2012
New Revision: 375074

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375074
Log:
Update sip_request_call SIP dial string documentation.

This was missed when merging review r1859.

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=375074&r1=375073&r2=375074
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Tue Oct 16 14:21:39 2012
@@ -26986,13 +26986,18 @@
 /*! \brief PBX interface function -build SIP pvt structure
  *	SIP calls initiated by the PBX arrive here.
  *
- * \verbatim	
- * 	SIP Dial string syntax
- *		SIP/exten at host!dnid
- *	or	SIP/host/exten!dnid
- *	or	SIP/host!dnid
+ * \verbatim
+ *	SIP Dial string syntax:
+ *		SIP/devicename
+ *	or	SIP/username at domain (SIP uri)
+ *	or	SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
+ *	or	SIP/devicename/extension
+ *	or	SIP/devicename/extension/IPorHost
+ *	or	SIP/username at domain//IPorHost
+ *	and there is an optional [!dnid] argument you can append to alter the
+ *	To: header.
  * \endverbatim
-*/
+ */
 static struct ast_channel *sip_request_call(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
 {
 	struct sip_pvt *p;