Update sip_request_call SIP dial string documentation.

This was missed when merging review r1859.
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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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375080 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c5de402..7227ddd 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -29429,13 +29429,18 @@ static int sip_devicestate(const char *data)
 /*! \brief PBX interface function -build SIP pvt structure
  *     SIP calls initiated by the PBX arrive here.
  *
- * \verbatim   
- *     SIP Dial string syntax
- *             SIP/exten@host!dnid
- *     or      SIP/host/exten!dnid
- *     or      SIP/host!dnid
+ * \verbatim
+ *     SIP Dial string syntax:
+ *             SIP/devicename
+ *     or      SIP/username@domain (SIP uri)
+ *     or      SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
+ *     or      SIP/devicename/extension
+ *     or      SIP/devicename/extension/IPorHost
+ *     or      SIP/username@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;