From: Automerge script <automerge@asterisk.org>
Date: Tue, 16 Oct 2012 20:24:24 +0000 (+0000)
Subject: Merged revisions 375078 via svnmerge from
X-Git-Tag: 10.11.0-digiumphones-rc1~29
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=f8b78847350bfdfba693dac94af389ea7c364226;p=thirdparty%2Fasterisk.git

Merged revisions 375078 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10

................
  r375078 | wdoekes | 2012-10-16 14:22:44 -0500 (Tue, 16 Oct 2012) | 7 lines

  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
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1fac7a3258..7550988fcd 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -28059,13 +28059,18 @@ static int sip_devicestate(void *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, void *data, int *cause)
 {
 	struct sip_pvt *p;