[asterisk-commits] wdoekes: trunk r375080 - in /trunk: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Oct 16 14:25:13 CDT 2012


Author: wdoekes
Date: Tue Oct 16 14:25:11 2012
New Revision: 375080

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=375080
Log:
Update sip_request_call SIP dial string documentation.

This was missed when merging review r1859.
........

Merged revisions 375074 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 375078 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 375079 from http://svn.asterisk.org/svn/asterisk/branches/11

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=375080&r1=375079&r2=375080
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Oct 16 14:25:11 2012
@@ -29429,13 +29429,18 @@
 /*! \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, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *dest, int *cause)
 {
 	struct sip_pvt *p;




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